Max no. of call-legs?
Hi, I've a 2811 with a ISDN PRI to the PSTN and a PVDM2-64. The router is configured for receiving and making calls and is working fine. The problem is I never seem to be able to make more than 16 calls on my PRI. Can anyone advise where i should look?
Perhaps your telco activated only 16 channels, check with them.
Similar Messages
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Max no of calls that can be queued in ICM
Where can I find for a given installation the max no of calls that can queued for given ICM system.
I looked at that but I think the answer is still not well defined. It talks about max call per second.That means that it can receive 300 max calls per second in ICM/IVR system. It does not say anything about how many concurrent calls can sit in the system and be queued.
For example on IVR side, we clearly know that it can support X no of max concurrent calls or whatever the figure may be based on ports. However for queued calls in ICM, it does not seem to very obvious. Unless I am missing something here. -
SIP reason = "This call leg has been replaced"
Hi All,
I am trying to troubleshoot an issue with calls being dropped, and have noticed that on some of my dropped calls I see the following in the BYE message:
ms-diagnostics: 10026;source="Lync.domain.local";reason="This call leg has been replaced";component="MediationServer"
ms-diagnostics-public: 10026;reason="This call leg has been replaced";component="MediationServer"
ms-endpoint-location-data: NetworkScope;ms-media-location-type=intranet
The call path is PSTN-->Gateway-->Lync-->Trusted application Server-->Agent
I cant seem to find much on what the "This call leg has been replaced" message means, if anyone can help me out it would be appreciated.
Cheers
JAll,
I have now had some time to analyse the problem further. I have produced a more detailed way of reviewing the data. By using a custom SQL view (see attached) in the LCSCDR DB I can
now connect Excel to the database and using Power Pivot quickly breakdown the data and look for some form of trend. Now I can see a the precise time, client versions and affected users. I have compared this information with the Syslog outputs on the Audio
Codes gateway.
It seems that all the problems I have generally occur when clients are connected to only one of my front end servers. There does not appear to be a similar number of problems on the other one.
Can anyone else please confirm if they too have an Audio Codes gateway when suffering with this issue?
I now have CU6 applied to all of the Lync infrastructure. This has not resolved the problem but has reduced the frequency which the errors are occurring.
Does anyone else have any further thoughts on this?
Thanks
Ben -
Every Since I Downloaded iOS5 O My IPhone My Volume Has Been Effected It's @ Max But Incoming Calls Come In @ A Minimum. When Taking A Call Thru Speaker Caller Sound Very Faint. Music Sounds Very Low Can Somebody Help Me ?
Hello ElleMalloy,
Sorry Im Just Responding Phone Has Been Downloading And Charging.
Well I Will Start Here,
I Tried Everything I Even Restarted My Settings Just So I Can See if That Would Work Because Someone Sujested It But What They Neglected To Say That Their Probrolem Was With A Ipad ? Anywho To Keep It Short I Went Online Logged A Complaint With Apple Made A Appointment With A Apple Store In My Area To Take The Phone There After Trying Everything (Resetting, changing setting etc..........) Nothin Worked Upon Taking The Phone To The Store They Swapped Out My Phone For A New One Volume On New Phone Is Perfect And I Only Had To Wait 40 Minutes -
How to set busy trigger and max number of calls for a sip phone in SRST
How is it possible to set the maximum number of calls and busy trigger for a line on a sip phone in SRST .There is no commands like dual-line or octo-line for the max-dn like under call-manager-fallback .
Sent from Cisco Technical Support iPad AppSo is the answer then that there is no way for a SIP phone in SRST mode to handle multiple calls? I have a site currently failed over in SRST mode, and I have noticed that the reception phone can only handle one call at a time. If I place a second call to it I get a busy signal.
-
CUCM 9.x with Gateway - Troubleshoot call leg
Guys,
We have a CUCM 9.x at a main site (Site-A), with a h.323 Gateway.
The h.323 Gateway have a Sip Dial-peer with another cisco pbx in another Country (Site-B).
In the past, people at Site-B used to called 816XXX to reach Site-B devices, and People in site-A can dial 83XXXX to reach people at site-B.
I know that some change occured in the last two weeks, since that it longer work for inter-site call.
I know that the IOS of Site-A has been updated to a 15.1 ( BUT we did add the IP address truested command to trust all PBX )
But still it's not working..
I did a debug voip dialpeer, and I would like to have your opinion
When user from Site B dial : 816999
It’s going well to our router at Bellevue, as expected
040331: 5d10h: //-1/856FEA4E80C4/DPM/dpAssociateIncomingPeerCore:
Calling Number=83644, Called Number=6999, Voice-Interface=0x0,
I see there is a successful match with dial-peer 6999
040341: 5d10h: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=6999
This dial-peer 6999 point to Call Manager
dial-peer voice 6999 voip
description Outgoing to CUCM
destination-pattern 6...
session target ipv4:10.AA.XZ.ZZ
dtmf-relay h245-alphanumeric
And one have a recommendation ?You will need to see in depth into the h225 and h245 signalling as well ass the CCAPI process..
You should do this
debug voip ccapi inout
debug h225 asn1
debug h245 asn1
To capture all of these logs..you need to properly setup your gateway as follows
service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit
Then..
<Enable debugs, then test again.>
<Enable session capture to txt file in terminal program.> (such as Putty)
then do the ff:
terminal length 0
show logging
You can attach the logs here in a text file -
Max no of calls - Busy Trigger
Hi,
I don't know if this is a standard settings or not?
When I put 6 to maximum nr and 2 to Busy trigger & when I am on the call and somebody trying to call me he receive busy and the call is remaining on
missed calls.
But when I put 6 to maximum nr and 1 to Busy trigger & when I am on the call and somebody trying to call me he receive busy and the call is not remaining on missed calls
What I want is when some body is on the call and he receive a second call, the person who trying to call to receive busy and the call to remain in missed calls.
I have CallM 8.6I have configured it some time with AndPhone (http://andtek.com/communications-products-calllist.html).
The Busy Call is forwarded to an CTI-RP which is controlled by andphone. Andphone evaluates the forwarding party and then forwards the call to a busy extension (e.g. an dial-peer on an h.323 gateway where busy signal is configured).
I'm sure there are other 3rd party solutions. -
Hi guys,
We have a UCCX 8.5 cluster and we want to set up a feature like IVR Rebound: caller talks to agent, and then agent can conference the caller to another IVR (internal or external IVR) for the caller to process the authorization, during the authorization process, agent must be on hold (cannot hear/interupt the authorization process), after the authorization is done (either successful or not), agent is connected to caller again.
Understand that it is not something nature to UCCX, I'm thinking of setting up system like this:
1. Caller calls to CallCenter and connect to agent.
2. Agent transfers (transfer, not conference) the call to an IVR script.
3. IVR accepts the call and put the call on hold. Now the caller is connected to IVR, agent is free.
4. IVR calls back to agent (as the agent will need to do something with the 2nd IVR) --> we have 3 party conference: IVR, agent, caller.
5. IVR makes another call to the 2nd IVR (can be external IVR, which is used for authorization) --> we have 4 party conference: IVR, agent, caller, IVR#2.
6. Agent key in some information needed by IVR#2.
7. IVR puts agent on hold and unhold the caller so the caller can proceed the authorization himself.
8. Authorization is done, IVR#2 disconnects, IVR needs to unhold agent so the agent and caller can talk to each other again.
My questions are:
Q1) In step 6 & 7: can IVR script detects a key sequence to determine when to start putting agent on hold. I'm thinking of Menu Grammar step but not sure if it is OK.
Q2) In step 8: can we catch the disconnection event (when IVR#2 disconnects) by using exception Channel Inactive Exception? I tried but when the IVR2 disconnects, the caller and agent cannot connect to each other but can only hear ringback tone/busy tone.
Thanks,
hoanghiepCCX has no ability to conference separate contacts together so this approach will not work. I can think of two options:
The agent transfers the caller to the authorization IVR at the conclusion of the call and checks for an approval status before processing the order. This is the cleanest approach from a CCX reporting perspective.
The agent transfers the caller to the external IVR which accepts an agent-specific identifier and transfers the caller back to the agent after authorization. You can use CCX or an external platform for this; the experience is essentially the same.
Please remember to rate helpful responses and identify helpful or correct answers. -
Unity Express - Incoming calls wont get voice mail
CUE works fine with telephones on my local network. Incoming and outgoing calls work fine.
However when I get an incoming call via SIP trunk the call will not get forwarded to unity express after 10 seconds. The line goes dead.
I searched for another post which suggested the following commands:
telephony-service
call-forward pattern .T
voice service voip
allow connections from h323 to sip
I've double checked them and there's still something wrong.
Here's my current configuration:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
telephony-service
load 7910 P00403020214
load 7960-7940 P00305000301
max-ephones 24
max-dn 24
ip source-address 192.168.20.1 port 2000
auto assign 1 to 24
system message Comtek
voicemail 3000
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
time-webedit
transfer-system full-consult
transfer-pattern 2...
transfer-pattern 3...
directory last-name-first
directory entry 2 2001 name Phone Two 7912
directory entry 3 2000 name Phone One 7970
ephone-dn 1 dual-line
number 2000 secondary 441833000000
call-forward busy 3000
call-forward noan 3000 timeout 10
no huntstop
ephone 1
no multicast-moh
device-security-mode none
mac-address 0017.0EF0.3642
type 7970
button 1:1
So pros, any suggestions?
ThanksI made a new dial-peer to handle incoming calls as follows.
dial-peer voice 1000 voip
description Incoming SIP
translation-profile incoming SIPin
voice-class codec 1
session protocol sipv2
incoming called-number .T
dtmf-relay rtp-nte
no vad
The translation-profile puts the call through to my 2000 extension.
This is my "show call active voice brief" when an external incoming call is ringing through to my 2000 ephone-dn.
To me this seems to show the dial-peer "1000" matching and using the g711ulaw codec
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1715 : 552 596706500ms.1 +-1 pid:1000 Answer +441833696807 connecting
dur 00:00:00 tx:0/0 rx:0/0
IP 87.127.240.98:16188 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
1715 : 553 596706510ms.1 +-1 pid:20001 Originate 2000 connecting
dur 00:00:00 tx:0/0 rx:0/0
Tele 50/0/1 (553) [50/0/1.0] tx:0/0/0ms None noise:0 acom:0 i/0:0/0 dBm
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
This is the "show call active voice brief" for an external incoming call when the call is established.
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1731 : 569 597220040ms.1 +3730 pid:1000 Answer +441833696807 active
dur 00:00:02 tx:105/16800 rx:104/16640
IP 87.127.240.98:15162 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
1731 : 570 597220060ms.1 +3700 pid:20001 Originate 2000 active
dur 00:00:02 tx:0/0 rx:105/16800
Tele 50/0/1 (570) [50/0/1.0] tx:16180/16180/0ms g711ulaw noise:0 acom:0 i/0:0/0 dBm
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
Not too sure where to go from here. -
Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP
Hi Cisco Community,
I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
Below is an example of a call that is connected with the current setup:
Note:
IP: 10.18.81.2 (CUBE)
IP: 10.18.81.11 (CUCM SUB)
IP: 10.111.111.254 (ITSP SBC)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 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, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
Session-Expires: 1800
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1417347869
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 301
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
s=SIP Call
c=I
PM-HO-VG-01#N IP4 10.18.81.2
t=0 0
m=audio 22256 RTP/AVP 18 0 8 101
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 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
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
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
a=ptime:20
Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf9
PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,application/xml
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 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
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
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
a=ptime:20
Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Type: application/sdp
Content-Length: 236
v=0
o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.80.40
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
PM-HO-VG-01#
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
PM-HO-VG-01#sh sip
PM-HO-VG-01#sh sip-ua call
PM-HO-VG-01#sh sip-ua calls
Total SIP call legs:2, User Agent Client:1, User Agent Server:1
SIP UAC CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 27218091323
Called Number : 0862000000
Bit Flags : 0xC04018 0x10000100 0x0
CC Call ID : 64511
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.111.111.254]:5060
Destn SIP Resp Addr:Port: [10.111.111.254]:5060
Destination Name : 10.111.111.254
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64511
Stream Type : voice+dtmf (0)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22256
Media Dest IP Addr:Port : [10.111.111.254]:20074
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 0218091323
Called Number : 0862000000
Bit Flags : 0xC0401E 0x10000100 0x80004
CC Call ID : 64510
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.18.81.11]:5060
Destn SIP Resp Addr:Port: [10.18.81.11]:5060
Destination Name : 10.18.81.11
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64510
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22350
Media Dest IP Addr:Port : [10.18.80.40]:21928
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1
PM-HO-VG-01#
PM-HO-VG-01#
PM-HO-VG-01#
As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 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: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22256 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 102 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 360
v=0
o=BroadWorks 316169737 2 IN IP4 10.111.111.254
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
a=inactive
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22350 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 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: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Length: 0
Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 103 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 306
v=0
o=BroadWorks 316169737 3 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 2
PM-HO-VG-01#00 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
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
a=ptime:20
Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 213
v=0
o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.81.10
t=0 0
m=audio 4000 RTP/AVP 18
a=X-cisco-media:umoh
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=sendonly
Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 101 BYE
Reason: Q.850;cause=86
P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 104 BYE
Reason: Q.850;cause=65
P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 Race Condition
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
Timestamp: 1417347889
CSeq: 104 BYE
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 200
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 101 BYE
Content-Length: 0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 86
Disconnect Cause (SIP) : 200
PM-HO-VG-01#Hi Manish,
Again, excellent feedback. Much appreciated.
I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
I will be doing some intensive test again later on this week and will send the logs.
Here is my question to both of you:
Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
Thanks again for your support fellows. -
Can't make outgoing call with Skype Connect
I have my Asterisk PBX configured with Skype Connect using SIP with TLS and SRTP. Most of my outgoing calls go through, but sometimes I can't get call out. I was able to leave asterisk console up and collect verbose and sip debug data. Can somebody help me diagnose why my calls aren't going through?
I've changed my external IP (I'm behind a NAT'd firewall) to 1.2.3.4 and my SIP profile's user ID to 11111111111111. and my domain name to test.com. If someone working for Skype needs that information they can email me and I'll send it privately.
My config:
[general]
context=default_context
allowguest=no
alwaysauthreject=yes
allowoverlap=no
udpbindaddr=0.0.0.0
tlsenable=yes
tlsbinddir=0.0.0.0
tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,tcp,tls
srvlookup=yes
dynamic_exclude_static = yes
buggymwi=yes
contactpermit=192.168.1.0/24
register => tls://111111111111111:[email protected]
[skype]
type=friend
context=from-skype
dtmfmode=rfc2833
host=sip.skype.com
username=11111111111111
fromuser=11111111111111
secret=abcd12345
disallow=all
allow=ulaw
allow=alaw
nat=yes
fromdomain=sip.skype.com
insecure=port,invite
transport=tls
srtpcapable=yes
encryption=yes
SIP Debugging enabled
[2012-08-23 19:22:33] NOTICE[16552]: chan_sip.c:13465 sip_reregister: -- Re-registration for [email protected]
> doing dnsmgr_lookup for 'sip.skype.com'
> ast_get_srv: SRV lookup for '_sips._tcp.sip.skype.com' mapped to host 1.sip.skype.com, port 5061
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 63.209.144.201:5061:
REGISTER sip:sip.skype.com:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as6edf93cf
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 32495 REGISTER
User-Agent: Asterisk PBX 10.5.2
Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:sip.skype.com:5061", nonce="5036b5770000182c78c1d1909cfd5c74e33f033c952d240d", response="81001ceacd91b16ebb956d3c55991471"
Expires: 120
Contact: <sip:[email protected]:5061;transport=TLS>
Content-Length: 0
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 200 OK
From: <sip:[email protected]>;tag=as6edf93cf
To: <sip:[email protected]>;tag=c990d13f-90f7a10d-0-55cb59a8-0
Call-ID: [email protected]
CSeq: 32495 REGISTER
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport=50541;received=1.2.3.4
Expires: 45
Contact: <sip:[email protected]:5061;transport=tls>;expires=45
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[2012-08-23 19:22:33] NOTICE[17932]: chan_sip.c:21399 handle_response_register: Outbound Registration: Expiry for sip.skype.com is 45 sec (Scheduling reregistration in 30 s)
<--- SIP read from UDP:192.168.1.16:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Scott's Office" <sip:[email protected]:5060>
Expires: 240
User-Agent: Cisco/SPA504G-7.5.2b
Content-Length: 234
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 88651316 88651316 IN IP4 192.168.1.16
s=-
c=IN IP4 192.168.1.16
t=0 0
m=audio 16484 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.1.16:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer 'scott_office' for 'scott_office' from 192.168.1.16:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.16:16484
Looking for 19739928881 in home (domain asterisk.test.com)
list_route: hop: <sip:[email protected]:5060>
<--- Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Executing [19739928881@home:1] Dial("SIP/scott_office-000000b0", "SIP/skype/+19739928881") in new stack
== Using SIP RTP CoS mark 5
Audio is at 9302
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 63.209.144.201:5061:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.5.2
Date: Thu, 23 Aug 2012 23:22:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 370
v=0
o=root 1671301052 1671301052 IN IP4 192.168.1.15
s=Asterisk PBX 10.5.2
c=IN IP4 192.168.1.15
t=0 0
m=audio 9302 RTP/SAVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
-- Called SIP/skype/+19739928881
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 100 Trying
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 407 Proxy Authentication Required
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip.skype.com", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", algorithm=MD5
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 63.209.144.201:5060
Transmitting (NAT) to 63.209.144.201:5061:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.5.2
Content-Length: 0
Audio is at 9302
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 63.209.144.201:5061:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.5.2
Proxy-Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:[email protected]", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", response="6efb0e37178bae868f0a1e0ddf110e3c"
Date: Thu, 23 Aug 2012 23:22:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 370
v=0
o=root 1671301052 1671301053 IN IP4 192.168.1.15
s=Asterisk PBX 10.5.2
c=IN IP4 192.168.1.15
t=0 0
m=audio 9302 RTP/SAVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 100 Trying
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: REGISTER
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 180 Ringing
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: SipGW 8
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
-- SIP/skype-000000b1 is ringing
<--- Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 408 Request Timeout
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[2012-08-23 19:22:45] WARNING[17932]: chan_sip.c:20947 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '[email protected]'. Giving up.
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 63.209.144.201:5060
Transmitting (NAT) to 63.209.144.201:5061:
ACK sip:[email protected]:5061;maddr=63.209.144.201;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.5.2
Content-Length: 0
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [19739928881@home:2] Hangup("SIP/scott_office-000000b0", "") in new stack
== Spawn extension (home, 19739928881, 2) exited non-zero on 'SIP/scott_office-000000b0'
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.1.16:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Scott's Office" <sip:[email protected]:5060>
User-Agent: Cisco/SPA504G-7.5.2b
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
Really destroying SIP dialog '[email protected]' Method: ACKI wound up calling skype support. This is the final sip.conf looks like. Hope it helps. Good luck.
Scott
[general]
context=default_context
allowguest=no
alwaysauthreject=yes
allowoverlap=no
udpbindaddr=0.0.0.0
tlsenable=yes
tlsbinddir=0.0.0.0
tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,tcp,tls
srvlookup=yes
dynamic_exclude_static = yes
buggymwi=yes
contactpermit=192.168.1.0/24
register => tls://[email protected]
[skype]
type=friend
context=from-skype
dtmfmode=rfc2833
host=sip.skype.com
username=user
fromuser=user
secret=pass
disallow=all
allow=ulaw
allow=alaw
nat=yes
fromdomain=sip.skype.com
insecure=port,invite
transport=tls
srtpcapable=yes
encryption=yes -
Show call history voice brief output
The output as below;
tok-fr-rtr>sh call hist voice brief
<ID>: <start>hs.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec>
MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
last <buf event time>s dur:<Min>/<Max>s
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
<codec> (payload size)
ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
<codec> (payload size)
Telephony <int> (callID) [channel_id] tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm
MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops> disc:<cause code>
speeds(bps): local <rx>/<tx> remote <rx>/<tx>
Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf>
bw: <req>/<act> codec: <audio>/<video>
tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
Telephony call-legs: 105
SIP call-legs: 0
H323 call-legs: 105
MGCP call-legs: 0
Total call-legs: 210
28CD : 232788573hs.5992 +162 +24465 pid:8712 Originate 8717898
dur 00:04:03 tx:11998/239960 rx:12151/243020 10 (normal call clearing (16))
IP 192.168.139.36:17424 rtt:247ms pl:237520/220ms lost:0/22/0 delay:50/50/70ms g729r8
28CD : 232788264hs.5993 +471 +24748 pid:46201 Answer
dur 00:04:02 tx:12139/242780 rx:11998/239960 10 (normal call clearing (16))
Telephony 3/0:0 (6048) [3/0.2] tx:235590/235590/0ms g729r8 noise:-82dBm acom:32dBm
1E6B : 232806587hs.5994 +417 +21065 pid:1 Answer
dur 00:03:26 tx:10447/208940 rx:10314/206280 10 (normal call clearing (16))
IP 192.168.139.104:19276 rtt:342ms pl:204470/10ms lost:0/0/0 delay:60/50/70ms g729r8
1E6B : 232806588hs.5995 +416 +21030 pid:46201 Originate 4622916
dur 00:03:26 tx:10297/205940 rx:10447/208940 10 (normal call clearing (16))
Telephony 3/0:0 (6051) [3/0.3] tx:207560/207560/0ms g729r8 noise:-82dBm acom:41dBm
Can someone tell me what does the value for pl:<play>/<gap>ms is refering to?
cheers!
csyeoanitachoi3,
You can use the below command to clear a voice call. Please exercise caution to ensure that this is not an on-going valid conversation.
look at the output of 'show call active voice brief' and grab the ID. Let's say you want to clear the call with the ID 7D4 in your example. Execute the below CLI:
clear call voice causecode 16 id 7D4
Execute 'show call active voice brief' to see if the call is cleared. -
Calls are not getting thru in Cisco voice GW for a particular Number
Cisco gateway is connecte to a PBX with an Qsig interface, for a particualr destination number the calls are not gettin estabilished.
the output of the Q931 debug :
Aug 16 16:17:46.145: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x7E05
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98396
Exclusive, Channel 22
Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
Facility i = 0x9FAA068001008201008B0100A11D0202010002010080144E455453202
F204C4F4E472044495354414E4345
Calling Party Number i = 0x2183, '8168911010'
Plan:ISDN, Type:National
Called Party Number i = 0x89, '18553808521'
Plan:Private, Type:Unknown
Sending Complete
Aug 16 16:17:46.149: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0xF
E05
Channel ID i = 0xA98396
Exclusive, Channel 22
Aug 16 16:17:55.709: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x
FE05
Cause i = 0x80BF - Service/option not available, unspecified
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x7E0
5
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref =
0xFE05
The Qsig and dial-peer configration :
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vadHi Raj,
My name is Edson Pineiro, I understand that your problem description is in regards to failed incoming calls from a qsig trunk.
According to the received q931 setup message I can see the called party number is 18553808521 and as so the gateway should route the dnis based on the best match in destination-pattern. My first suggestion would be to ensure your outgoing dial-peers has a matching destination-pattern that matches the dialed number, for example:
dial-peer voice 1 voip
destination-pattern 1T
The T is a wild card for any digit any length
Or you can be very specific.
dial-peer voice 1 voip
destinaton-pattern 18553808521
The next suggestion would be to ensure that your incoming pots dial-peers contains 'direct-inward dial'. This is so that you don't receive secondary dial tone when dialing in, which I don't think is happening here.
Another suggestion would be to remove 'isdn overlap-receiving' from interface serial 0/0/0:23. Reason being is that the DNIS received is enbloc and not overlapping. You can clearly see that the complete e164 number is received within the setup and no further digits are needed.
But overall the disconnect cause code is 0x80BF the 80 portion is related to the source of the disconnect which is the router and BF "Service/option not available, unspecified" which is described as:
The network or remote equipment cannot provide the service option that the user requests, due to an unspecified reason. A subscription problem can cause this issue.
Any ways seems like the router does not support the protocol or type of message included in the Setup. After decoding one of the facility message:
Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
decode -->
Facility IE first byte (protocol profile): 0x9f(Network Extentions), depends on Network Protocol Profile
**Note:
**0x91/0x9f both be used by older qsig spec, including:
**ISO 11582:1995, ETSI 300 239:1993/1995
**newer qsig spec use 0x9f only, including:
**ISO 11582:1995/Cor.1:1999, ECMA 165(4th), ETSI 300 239:2003
**see CSCeb58118 for CCM compatibility issue
NetworkFacilityExtension ::= {
sourceEntity: 0
destinationEntity: 0
NetworkProtocolProfile not present
APDU is a ROSE
0
DivertingLegInformation2Invoke ::= {
invokeID: 1793
operationValue: 21
argument: DivertingLegInformation2Arg ::= {
diversionCounter: 1
diversionReason: 1
originalDiversionReason: 1
divertingNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
originalCalledNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
redirectingName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
originalCalledName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
Looks like this is a redirected call (call forward or transfer), the redireted number is "50005998" and the other end of the PRI maybe attempting to do either a 2 B channel transfer or B channel optimization, which is not supported certain gateways or needs the use of a tcl scripts. Any ways is it possible to confirm if such features are enabled on the other end of the qsig trunk? and what the number 50005998 is assigned too. This may warrant a TAC case.
However please ensure your carry through the first three configuration changes before looking at the possible bad facility message.
Here are some good documents on ISDN, IOS dial-peers and call legs:
Understanding debug isdn q931 Disconnect Cause Codes
http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a008012e95f.shtml
Configuring Telephony Call-Redirect Features
Two B-Channel Transfer
http://www.cisco.com/en/US/docs/ios/voice/ivr/pre12.3_14_t/configuration/guide/ivrapp.pdf
Understanding Dial Peers and Call Legs on Cisco IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
Understanding Direct-Inward-Dial (DID) on IOS Voice Digital (T1/E1) Interfaces
http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_note09186a00801142f8.shtml
Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#prereq
Voice Translation Rules
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml
Let me know how you go.
Thanks again for asking the tuff questions.
Cheers
Edson -
Dial-Peer matches but fails to call out
Hello,
Am trying to get my CME configured for Callcentric. I have both an inbound and an outbound plan.
With my dial-peers configured for standard 11-digit and 10-digit dialing, calls go to fast busy after all digits except the last two are dialed. Debug shows a dial-peer match initially, then states no match and the call fails. If I change the destination pattern to match my cell phone number exactly, I can dial all the digits but the call still fails. Anyone have a suggestion?
Here are my dial peers:
dial-peer voice 700 voip
description SIP Trunk - Incoming
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:callcentric.com
incoming called-number .%
dial-peer voice 701 voip
description SIP Trunk - Outgoing 3-Digit Calls
translation-profile outgoing SIP_1
preference 1
destination-pattern 9[2-8]11
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
dial-peer voice 702 voip
description SIP Trunk - Outgoing 11-Digit Calls
translation-profile outgoing SIP_1
preference 1
destination-pattern 91[2-9].......
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
dial-peer voice 703 voip
description SIP Trunk - Outgoing 10-Digit Calls
translation-profile outgoing SIP_1
preference 1
destination-pattern 9[2-9].......
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
no vad
And here is the debug associated with a call:
*Dec 26 22:26:27.854: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=7018, Called Number=, Voice-Interface=0x4A4AE7B0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:27.854: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20009
GMIT-VOICEROUTER01#
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:29.526: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=91, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=91
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:29.914: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Numbe
GMIT-VOICEROUTr=, Called Number=912, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=912
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:30.174: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:30.514: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:30.822: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=91207, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:30.822: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=91207
*Dec 26 22:26:30.826: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:30.826: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:31.342: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=912072, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:31.342: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=912072
*Dec 26 22:26:31.346: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:31.346: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:31.542: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722
*Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:31.546: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=91207227, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=91207227
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:31.934: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=912072277, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=912072277
*Dec 26 22:26:32.602: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Dec 26 22:26:32.606: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.382: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.386: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=9120722776, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=9120722776, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=702
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.390: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9120722776, Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9120722776
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Dec 26 22:26:33.394: //-1/98A189FD81D3/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=702
*Dec 26 22:26:33.574: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=91[2-9]......., Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:33.578: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
*Dec 26 22:26:36.374: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=7018$, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Dec 26 22:26:36.378: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules AttemptER01#Translation profile:
voice translation-rule 3
rule 1 /^7../ /2072267262/
voice translation-rule 4
rule 1 /^9\(1....\)/ /\1/
rule 2 /^9207\(...\)/ /\1/
rule 3 /^9\(011.*\)/ /\1/
rule 4 /^9\([2-9]11\)/ /\1/
voice translation-profile SIP_1
translate calling 3
translate called 4
Here is debug ccsip messages:
*Dec 27 14:10:16.598: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 207.5.178.214:5060;branch=z9hG4bKd77b793048igqgkfd0g1.1
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
Max-Forwards: 69
Call-ID: SDp6vve01-196593d11e4bf68c71f8a4085d7de7d0-c54gcb0
From: ;tag=SDp6vve01-callagent.gwi.net+1+8bfe3a+d3cf6d84
CSeq: 938331054 OPTIONS
Organization: MetaSwitch
Supported: resource-priority, 100rel
Content-Length: 0
Contact:
To:
*Dec 27 14:10:16.606: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.5.178.214:5060;branch=z9hG4bKd77b793048igqgkfd0g1.1
From: ;tag=SDp6vve01-callagent.gwi.net+1+8bfe3a+d3cf6d84
To:
GMIT-VOICEROUT166>;tag=F1B5120-18BD
Date: Fri, 27 Dec 2013 14:10:16 GMT
Call-ID: SDp6vve01-196593d11e4bf68c71f8a4085d7de7d0-c54gcb0
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 938331054 OPTIONS
Supported: 100rel,resource-priority,replaces,sdp-anat
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 172
v=0
o=CiscoSystemsSIP-GW-UserAgent 4484 7548 IN IP4 66.55.220.166
s=SIP Call
c=IN IP4 66.55.220.166
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 66.55.220.166
ER01#
GMIT-VOICEROUTER01#
*Dec 27 14:10:34.834: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2066961728-1849102819-2185007278-567139419
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, B
GMIT-VOICEROUTER01#YE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1388153434
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 297
v=0
o=CiscoSystemsSIP-GW-UserAgent 3022 9963 IN IP4 66.55.220.166
s=SIP Call
c=IN IP4 66.55.220.166
t=0 0
m=audio 19258 RTP/AVP 18 101 19
c=IN IP4 66.55.220.166
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
*Dec 27 14:10:34.906: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
f: "Server Room" [email protected]>;tag=F1B9854-8A5
t: [email protected]>
i: [email protected]
CSeq: 1
GMIT-VOICEROUT01 INVITE
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="3755ae79fd668c2035ebb90cdc12d030", opaque="", stale=TRUE, algorithm=MD5
l: 0
*Dec 27 14:10:34.914: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK5795232C
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
*Dec 27 14:10:34.914: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2066961728-1849102819-2185007278-567139419
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1388153434
Contact:
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="17772882353",realm="callcentric.com",uri="sip:[email protected]:5080",response="cbac03a76a23b6a35ebbee966c00a577",nonce="3755ae79fd668c2035ebb90cdc12d030",opaque="",algorithm=MD5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 297
v=0
o=CiscoSystemsSIP-GW-UserAgent 3022 9963 IN IP4 66.55.220.166
s=SIP Call
c=IN IP4 66.55.220.166
t=0 0
m=audio 19258 RTP/AVP 18 101 19
c=IN IP4 66.55.220.166
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
*Dec 27 14:10:34.990: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Incorrect Authentication
v: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
f: "Server Room" [email protected]>;tag=F1B9854-8A5
t: [email protected]>
i: [email protected]
CSeq: 102 INVITE
l: 0
*Dec 27 14:10:35.002: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP 66.55.220.166:5060;branch=z9hG4bK579619F9
From: "Server Room" [email protected]>;tag=F1B9854-8A5
To: [email protected]>
Date: Fri, 27 Dec 2013 14:10:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Here is debug voip ccapi inout:
GMIT-VOICEROUTER01#debug voip ccapi inout
voip ccapi inout debugging is on
GMIT-VOICEROUTER01#
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=7018
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-lastrdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/cc_api_call_setup_ind_common:
Interface=0x4A4AE7B0, Call Info(
Calling Number=7018,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE,
Incoming Dial-peer=20009, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
GMIT-VOICEROUT, Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/ccCheckClipClir:
In: Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:55.326: //-1/8912F77B8243/CCAPI/ccCheckClipClir:
Out: Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:55.326: :cc_get_feature_vsa malloc success
*Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:55.326: cc_get_feature_vsa count is 1
*Dec 27 14:10:55.326: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:55.326: :FEATURE_VSA attributes are: feature_name:0,feature_time:1282234808,feature_id:151
*Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=7018(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown))
*Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/cc_process_call_setup_ind:
Event=0x49A103B8
*Dec 27 14:10:55.330: //12898/8912F77B8243/CCAPI/ccCallSetContext:
Context=0x4C5A319C
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 12898 with tag 20009 to app "_ManagedAppProcess_Default"
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccCallSetupAck:
Call Id=12898
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/cc_api_set_transfer_info:
Transfer Number=, Transfer Reason=0x0
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=TRUE, Tone=Dial Tone,
Tone Direction=Network, Params=0x0, Call Id=12898
*Dec 27 14:10:55.334: //12898/8912F77B8243/CCAPI/ccSetDigitTimeouts:
Initial Digit Timeout=-1000(ms), Inter Digit Timeout=-1000(ms)
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/ccSetDigitTimeouts:
Call Entry(Inter Digit Timeout=10000(ms), Initial Digit Timeout=10000(ms))
*Dec 27 14:10:55.338: //12898/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x3262, digit_event=0x1, enable=TRUE, consume=FALSE)
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=12898
*Dec 27 14:10:55.338: //12898/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x4A4AE7B0, callID=0x3262, disp=0, digit_event=0x1, enable=TRUE, consume=FALSE)
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x4A4AE7B0, Call Id=12898
*Dec 27 14:10:55.338: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=15000(ms), Inter Digit Timeout=10000(ms))
*Dec 27 14:10:56.650: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=9, DigitBeginFlags=0x0,
Rtp Timestamp=0x9D41D0, Rtp Expiration=0x0
*Dec 27 14:10:56.654: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=9, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:56.654: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:56.970: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=1, DigitBeginFlags=0x0,
Rtp Timestamp=0x9DBED0, Rtp Expiration=0x0
*Dec 27 14:10:56.974: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=1, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:56.974: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:57.290: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
Rtp Timestamp=0x9E3BD0, Rtp Expiration=0x0
*Dec 27 14:10:57.294: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:57.294: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:57.610: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=0, DigitBeginFlags=0x0,
Rtp Timestamp=0x9EB8D0, Rtp Expiration=0x0
*Dec 27 14:10:57.614: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=0, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:57.614: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:57.890: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
Rtp Timestamp=0x9F35D0, Rtp Expiration=0x0
*Dec 27 14:10:57.894: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:57.894: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
Rtp Timestamp=0x9FB2D0, Rtp Expiration=0x0
*Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.162: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, DigitBeginFlags=0x0,
Rtp Timestamp=0xA02FD0, Rtp Expiration=0x0
*Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=2, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.314: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
Rtp Timestamp=0xA0ACD0, Rtp Expiration=0x0
*Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.582: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, DigitBeginFlags=0x0,
Rtp Timestamp=0xA129D0, Rtp Expiration=0x0
*Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=7, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:58.754: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_begin:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=6, DigitBeginFlags=0x0,
Rtp Timestamp=0xA1A6D0, Rtp Expiration=0x0
*Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Destination Interface=0x0, Destination Mask=0x3, Destination Call Id=-1,
Source Call Id=12898, Digit=6, Duration=100,
Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone Mode=DTMF
*Dec 27 14:10:59.022: //12898/8912F77B8243/CCAPI/cc_api_call_digit_end:
Call Entry(Handoff Depth=0)
*Dec 27 14:10:59.026: //12898/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x3262, digit_event=0x0, enable=FALSE, consume=FALSE)
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=12898
*Dec 27 14:10:59.026: //12898/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x4A4AE7B0, callID=0x3262, disp=0, digit_event=0x0, enable=FALSE, consume=FALSE)
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x4A4AE7B0, Call Id=12898
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=15000(ms), Inter Digit Timeout=10000(ms))
*Dec 27 14:10:59.026: //12898/8912F77B8243/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=702, Params=0x4C5A0BDC, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCheckClipClir:
In: Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCheckClipClir:
Out: Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
Destination Pattern=91[2-9]......., Called Number=120722776, Digit Strip=FALSE
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/ccCallSetupRequest:
Calling Number=20722672628(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=120722776(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Server Room
Account Number=, Final Destination Flag=FALSE,
Guid=8912F77B-6E37-11E3-8243-90AE21CDDC5B, Outgoing Dial-peer=702
*Dec 27 14:10:59.030: //12898/8912F77B8243/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=20722672628
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=120722776
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-lastrdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
*Dec 27 14:10:59.034: //12898/8912F77B8243/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x48C27BD0, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=20722672628,(Calling Name=Server Room)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=120722776(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=702, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
*Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:59.034: :cc_get_feature_vsa malloc success
*Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:59.034: cc_get_feature_vsa count is 2
*Dec 27 14:10:59.034: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Dec 27 14:10:59.034: :FEATURE_VSA attributes are: feature_name:0,feature_time:1282234584,feature_id:152
*Dec 27 14:10:59.034: //12899/8912F77B8243/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
*Dec 27 14:10:59.034: //12899/8912F77B8243/CCAPI/ccCallSetContext:
Context=0x4C5A0B8C
*Dec 27 14:10:59.034: //12898/8912F77B8243/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=702
*Dec 27 14:10:59.038: //12899/8912F77B8243/CCAPI/cc_api_call_proceeding:
Interface=0x48C27BD0, Progress Indication=NULL(0)
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/cc_api_call_disconnected:
Cause Value=57, Interface=0x48C27BD0, Call Id=12899
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=57, Retry Count=0)
*Dec 27 14:10:59.270: //12898/xxxxxxxxxxxx/CCAPI/ccCallReleaseResources:
release reserved xcoding resource.
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallSetAAA_Accounting:
Accounting=0, Call Id=12899
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallDisconnect:
Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=57)
*Dec 27 14:10:59.270: //12899/8912F77B8243/CCAPI/ccCallDisconnect:
Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
*Dec 27 14:10:59.274: //12899/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x48C27BD0, Tag=0x0, Call Id=12899,
Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
*Dec 27 14:10:59.274: //12899/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
*Dec 27 14:10:59.274: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:10:59.274: :cc_free_feature_vsa freeing 4C6D58D0
*Dec 27 14:10:59.274: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:10:59.274: vsacount in free is 1
*Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/ccCallDisconnect:
Cause Value=57, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/ccCallDisconnect:
Cause Value=57, Call Entry(Responsed=TRUE, Cause Value=57)
*Dec 27 14:10:59.278: //12898/8912F77B8243/CCAPI/cc_api_get_transfer_info:
Transfer Number Is Null
*Dec 27 14:11:02.250: //12898/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x4A4AE7B0, Tag=0x0, Call Id=12898,
Call Entry(Disconnect Cause=57, Voice Class Cause Code=0, Retry Count=0)
*Dec 27 14:11:02.250: //12898/8912F77B8243/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
*Dec 27 14:11:02.250: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:11:02.250: :cc_free_feature_vsa freeing 4C6D59B0
*Dec 27 14:11:02.250: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Dec 27 14:11:02.250: vsacount in free is 0ER01# -
Cisco Jabber for Windows in Extend and Connect mode and making outbound calls
Hi guys,
I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
Has anyone got this working or can provide some guidance?
Thanks.Hi guys,
I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
Has anyone got this working or can provide some guidance?
Thanks.
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