SIP trunk early offer

Unanswered Question
Jun 14th, 2012

Hi,

We have configured a trunk with a provider using CUBE.

Callmanager--------CUBE--------Provider

The provider wants early offer and G729r8

So we configured a SIP trunk with a device pool/region so that only g729 is allowed between SIP trunk and the rest.

We have configured an IOS MTP resource, and this is registered on callmanager:

!

dspfarm profile 2 mtp

codec g729r8

maximum sessions software 20

associate application SCCP

!

!

The mtp resource is assign to the trunk using MR-list and MR-group

The trunk has MTP enabled with "MTP prefered Codec" G729b/G729ab

On CUBE we enable "deb ccsip mess" and we see the invite comming from callmanager, but without attached SDP

What must be done to make callmanager use early-offer?

Thanks for the help,

Jan

I have this problem too.
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Joseph Martini Thu, 06/14/2012 - 15:01

Check MTP required on the SIP trunk configuration page or if you are on CUCM 8.5 or later go to the SIP Profile (Device > Device Settings > SIP Profile) and check "Early Offer support for voice and video calls (insert MTP if needed)".

j.huizinga Thu, 06/14/2012 - 22:23

Hello Joe,

I have MTP required enabled on the trunk, but it doesn't really change anything.

If I enable, or don't enable "MTP required" the SIP INVITE from callmanager is the same.

We are running Callmanager 7.1.5, so we don't have the option in the SIP profile.

When we establish a call I can see that the IOS MTP is involved (show sccp connections) but no early offer.

To make a successfull cal we force the provider dial-peer to use G729 only.

Unfortunately this IOS doesn't have the forced early offer option.

Is there someone who has succeeded in making CM 7.1.5 do an early offer?

Thanks,

Jan

nikshah Fri, 06/15/2012 - 00:07

Hi Jan,

Here is your answer

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/media.html#wp1055833

Snippet from above

SIP Early Offer

SIP negotiates media exchange via Session Description Protocol (SDP),  where one side offers a set of capabilities to which the other side  answers, thus converging on a set of media characteristics. SIP allows  the initial offer to be sent either by the caller in the initial INVITE  message or, if the caller chooses not to, the called party can send the  initial offer in the first reliable response. By default, Unified CM  sends the INVITE without an initial offer, and it requires MTP resources  to send the offer in the INVITE. Note that this initial offer is  limited to the G.711 codec only.

Also note that MTP resources are not required for incoming INVITE messages, whether or not they contain an initial offer.

Regards

Niket

j.huizinga Fri, 06/15/2012 - 04:06

Hi Niket,

You seem to be right, when I force 711 I see early offer.

But it is weird that on the trunk configuration page, it allows me to select a g729 codec as prefered codec.

Can you explain why this option is there, although it doesn't sem to work?

Thanks for your help,

Jan

nikshah Fri, 06/15/2012 - 14:34

Hi Jan,

I dont know whats the logic and i dont have a confirmed answer, behind not supporting the g729 codec in CM 7.1.5 , however when i open the SRND for CM 8.x

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1126420

I dont see any requirements on the codec.

So somewhere between CM 7.x and CM 8.x there must be a fix for supporting the g729 codec on early offer.

Hope this helps.

Regards

Niket

j.huizinga Sun, 06/17/2012 - 11:56

Hi Niket,

Thank you very much.

We are testing now with another router that has an IOS which supports 'SIP early offer forced' and we hope that this will work.

So the callmanager will send a normal INVITE (delayed offer) and the CUBE will send out a early offer to the provider.

Regards,

Jan

Ayodeji oladipo... Mon, 06/18/2012 - 00:51

J,

Did the early offer-forced work?

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j.huizinga Mon, 06/18/2012 - 03:10

Hi,

We haven't configured yet. I shall let the community know.

Thanks for all,

Jan

Ayodeji oladipo... Tue, 06/19/2012 - 00:26

J,

I just saw your query on the ask the expert forum. I thought I should chip in here for you.

Yes CUCM can send delay offer and CUBE can send early offer. This is how we have it configured for one of my biggest customers with 3 clusters (8 server/cluster) and it works very well...

Here is a snippet from our CUBE config..

sip

early-offer forced.

Here is a sample trace..

+++++CUCM sends invite without early offer+++++++++

Received:
INVITE sip:07544455678@172.16.10.74:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.174:5060;branch=z9hG4bKa41e64e002602
From: "Chelmsford" ;tag=517285~ffa80926-5fac-4dd6-b405-2dbbc56ae9a2-400694295
To:
Date: Mon, 18 Jun 2012 09:46:00 GMT
Call-ID: 67d44180-fde1f8d8-3aead-ae28690a@172.16.10.174
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1741963648-0000065536-0000064326-2921883914
Session-Expires:  84600
Contact:
Max-Forwards: 70
Content-Length: 0

++++++++++++++CUBE sends a trying to CUCM++++++++++++++++

Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.10.174:5060;branch=z9hG4bKa41e64e002602
From: "Chelmsford" ;tag=517285~ffa80926-5fac-4dd6-b405-2dbbc56ae9a2-400694295
To:
Date: Mon, 18 Jun 2012 09:46:00 GMT
Call-ID: 67d44180-fde1f8d8-3aead-ae28690a@172.16.10.174
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

+++++++++CUBE sends an invite to ITSP with Early Offer++++++++++


010625: Jun 18 09:46:00.371: //452987/67D441800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:07544455678@10.100.33.54:5070 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.74:5060;branch=z9hG4bKFE71F2421
Remote-Party-ID: "Chelmsford" ;party=calling;screen=no;privacy=off
From: "Chelmsford" ;tag=88EFEFBA-14A2
To:
Date: Mon, 18 Jun 2012 09:46:00 GMT
Call-ID: 3EDB1C4A-B86111E1-94E48F4D-5D7E5E41@172.16.10.74
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1741963648-0000065536-0000064326-2921883914
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1340012760
Contact:
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires:  84600
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 354

v=0
o=CiscoSystemsSIP-GW-UserAgent 6030 604 IN IP4 172.16.10.74
s=SIP Call
c=IN IP4 172.16.10.74
t=0 0
m=audio 18110 RTP/AVP 18 0 8 100 101
c=IN IP4 172.16.10.74
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

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j.huizinga Tue, 06/19/2012 - 00:46

Thank you very much!

We are going to configure it like this.

Unfortunately one of our CUBE router has not the forced SIP early offer command, and we need to install more flash and memory and then do an IOS upgrade.

We now make a test with a 2811 that has the correct IOS, but this router has such a weird issue that I think I have to open a TAC case. In this router I can not apply the "bind control source-interface" command. I can type it, but the"show run" doesn't show it. So weird.

We did format flash, install another IOS, but it doesn't apply the bind commands.

Thanks!

Jan

Ayodeji oladipo... Tue, 06/19/2012 - 00:52

How did you apply the command..Is it like this..

sip

  bind control source-interface gig0/0

  bind media source-interface gig0/0

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Ayodeji oladipo... Tue, 06/19/2012 - 00:56

What IOS version are you running?

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j.huizinga Tue, 06/19/2012 - 00:59

c2800nm-ipvoice_ivs-mz.124-20.T6.bin

c2800nm-adventerprisek9-mz.124-24.T7.bin

c2800nm-adventerprisek9-mz.124-24.T4.bin

c2800nm-adventerprisek9-mz.124-24.T5.bin

Pick one

We tried all of them.

Thanks,

Jan

Ayodeji oladipo... Tue, 06/19/2012 - 01:37

Can you send the output of this command..

show sip-ua status

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j.huizinga Tue, 06/19/2012 - 01:50

Hi,

At this moment I have removed this router from customer site, and it is in my office.

We got crazy about this router not accepting the bind commands.

But regardless of configuration, the bind commands should be accepted I think.

Thanks,

Jan

jfernandorosa Mon, 09/02/2013 - 11:41

Hi Aokanlawon,

I have the similar problem:

My scenario:

CUCM 9X -->SIP TRUNK--> CUBE --> ISP SIP

I need invite Early Offer to ISP, for DTMF problems, I don´t like the use CUCM fot this.

I set in CUBE (early-offer forced), but if I removed pass-thru content sdp, i received fas busy and CUCM return Internal Server Error:

sip

  bind control source-interface GigabitEthernet0/1

  bind media source-interface GigabitEthernet0/1

  early-offer forced

  midcall-signaling passthru

  pass-thru headers unsupp

  no call service stop

how can I solve this?

Thanks!

Joao

Ayodeji oladipo... Mon, 09/02/2013 - 12:36

Can you do a test call and send us "debug ccsip messages" attach it here

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Ayodeji oladipo... Mon, 09/02/2013 - 12:54

The log you attached was a succesful call and your cube didnt send EO to your ITSP. I didnt see any error in the log

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jfernandorosa Mon, 09/02/2013 - 13:41

In this case not have problem, is correct.

But in some calls the ISP don´t invite SDP payload with DTMF information (telephone event) and DTMF fails in this case.

I attached log problem.

Thanks.

Joao

Ayodeji oladipo... Mon, 09/02/2013 - 14:30

There is nothing you can do, if your ITSP doesnt advertise any DTMF capabilites in their SDP. You need to contact them and have it corrected. CUBE can only respond to what is offered. This is a problem with them so get them to sort it out

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jfernandorosa Mon, 09/02/2013 - 15:02

Hi.

I remove the SIP configs below for invite SDP EO to ISP.

sip

  pass-thru headers unsupp

  pass-thru content sdp

  no call service stop

and use (early-offer forced)

The ISP response with payload complete in this case, I dialed the same number with problems, but my call rinring and return a fast busy in this case, CUBE return internal server error for CUCM.

Sent:

SIP/2.0 500 Internal Server Error

Via: SIP/2.0/TCP 21.10.0.7:5060;branch=z9hG4bK13eb293705d2

From: "ATA187 Core" ;tag=21576~fb89236f-816b-47f5-8c94-b8d3c388dd7c-64665064

To: ;tag=3E21E738-DD8

Date: Mon, 02 Sep 2013 21:44:24 GMT

Call-ID: 263d9280-22510742-b70-7000a15@21.10.0.7

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.2.4.M3

Reason: Q.850;cause=96

Content-Length: 0



Ayodeji oladipo... Tue, 09/03/2013 - 01:45

You need to post the full debug, for us to know whats happening. Cause code 96 means that a madatory IE is missing

Typical scenarios include:

  • Mandatory Contact field missing in SIP message.
  • Session Description Protocol (SDP) body is missing.

So until I see the full log, I wont know what is wrong

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jfernandorosa Tue, 09/03/2013 - 07:00

Hi,

Attached logs with error 96 for you analise.

I don´t see the SDP payloads.

Thanks for help.

Joao

jbollen Tue, 09/03/2013 - 07:50

Well, in this case (error code 96) the service provider is:

a. removig the SDP in the subsequent 180 Ringing

b. not including an SDP in the 200 OK

That's why the router spits out cause code 96.

Talk to the Service Provider, I would suggest...

cheers,

Jan

Mukesh Kumar Mon, 08/24/2015 - 05:31

Hi Ayodeji,

We have the following setup:

Phones-- PBX--- Voice gateway --- SIP provider 

 

When we try to make any outbound call, call gets connected and when the destination end receives the call the call gets disconnected. 

It shows the cause 16 of disconnecting the call.

As I have searched for it, I found that call is cleared normally. But we haven't cleared call.

When I have checked the debugs on Voice Gateway. I have found that when we make any outboud call then after registration, we receive 183 Session progress message from the SIP provider. After that SDP message send from our end.

It looks like that the SIP provider is using early offer and on our end delay offer is running.

Can you please tell me if one voice gateway is using delay offer and the SIP provider will be using early offer then what will happen?

Does the call be successful or it has got connected?

 

Regards;

MUKESH KUMAR | Network Engineer
Spooster IT Services
Computer Networking Solutions

 

 

 

Vivek Batra Mon, 08/24/2015 - 05:54

Which PBX?

Which voice gateway?

When I have checked the debugs on Voice Gateway. I have found that when we make any outboud call then after registration, we receive 183 Session progress message from the SIP provider. After that SDP message send from our end.

It looks like that the SIP provider is using early offer and on our end delay offer is running.

It's always UAC who can decide whether to use early or delayed offer. So in your case, when you make an outgoing call, it's your gateway who can choose between early and delayed offer, not your service provider. It's true vice-versa when gateway receives incoming call from service provider, your service provider decides whether to use early or delayed offer.

Coming back to your question and comments, we can't say it was early or delayed offer as you've not mentioned whether INVITE from gateway was sent with or without SDP. If it was with SDP, gateway is using early offer and if it was without SDP, gateway is using delayed offer.

Can you please tell me if one voice gateway is using delay offer and the SIP provider will be using early offer then what will happen?

The question is not much relevant because UAS has to respond as per request revived from UAC. If UAC has initiated call using delayed offer, UAS must have to support delayed offer else UAS should reject the call if doesn't support delayed offer.

Mukesh Kumar Tue, 08/25/2015 - 00:31

Thanks for the information Vivek.

In our case call is disconnected from our end.

When the call gets connected and the destination receives the call then from our end "bye" is sent to the SIP provider.

Is this is related to SIP early offer or SIP early delay?

Calling number is on our end and the called number is on the SIP provider end.

 

Regards,

MUKESH KUMAR | Network Engineer
Spooster IT Services
Computer Networking Solutions
Mukesh Kumar Tue, 08/25/2015 - 00:46

Hello Vivek,

Below is the config:

 

version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname Cisco
!
boot-start-marker
boot-end-marker
!
!
card type t1 0 0
!
no aaa new-model
!

network-clock-participate wic 0 
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
!
ip name-server 8.8.8.8
ip name-server 4.2.2.2
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
isdn switch-type primary-ni
!
!
!
voice service voip
 ip address trusted list
  ipv4 70.70.1.2
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
!
voice class codec 1
 codec preference 1 g711ulaw
!
!
!
!
voice translation-rule 11
 rule 1 /^\(.......\)$/ /500\1/
!
voice translation-rule 12
 rule 1 /^.*/ /8547845654/
!
!
voice translation-profile SIP-CALL
 translate calling 12
 translate called 11
!
!
voice-card 0
 dsp services dspfarm
!
crypto pki token default removal timeout 0
!
!
!
!
!
redundancy
!
!
controller T1 0/0/0
 clock source internal
 cablelength long 0db
 pri-group timeslots 1-24
!
!
!
interface GigabitEthernet0/0
 ip address 40.40.1.5 255.255.255.252
 ip flow ingress
 ip flow egress
 duplex full
 speed 100
!
!
interface Serial0/0/0:23
 no ip address
 ip flow ingress
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn protocol-emulate network
 isdn incoming-voice voice
 isdn outgoing display-ie
 isdn outgoing ie redirecting-number 
 no cdp enable
!
ip forward-protocol nd
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 40.40.1.6
!

!
control-plane
!
!
voice-port 0/0/0:23
!
!
!
mgcp profile default
!
!
dial-peer voice 100 pots
 incoming called-number .
 direct-inward-dial
!
dial-peer voice 20 voip
 translation-profile outgoing SIP-CALL
 preference 3
 destination-pattern .T
 session protocol sipv2
 session target ipv4:70.70.1.2
 session transport udp
 incoming called-number .
 voice-class codec 1  
 dtmf-relay sip-notify rtp-nte
 no vad
!
dial-peer voice 10 pots
 destination-pattern 8547845655$
 forward-digits 4
 port 0/0/0:23
!
dial-peer voice 11 pots
 destination-pattern 8547845656$
 forward-digits 4
 port 0/0/0:23
!

!
sip-ua 
 credentials username 8547845654 password Cisco realm abc.sip.com
 authentication username 8547845654 password Cisco realm abc.sip.com
 retry invite 3
 retry register 3
 timers register 150
 registrar dns:abc.sip.com expires 200
 sip-server dns:abc.sip.com
!
!
!
!
line con 0
 login local
line aux 0
 login local
line vty 0 4
 login local
 transport input all
!
scheduler allocate 20000 1000
end

=====================================================

show controllers t1
T1 0/0/0 is up.
  Applique type is Channelized T1
  Cablelength is long 0db
  No alarms detected.
  alarm-trigger is not set
  Soaking time: 3, Clearance time: 10
  AIS State:Clear  LOS State:Clear  LOF State:Clear 
  Version info Firmware: 20100222, FPGA: 13, spm_count = 0
  Framing is ESF, Line Code is B8ZS, Clock Source is Internal.
  CRC Threshold is 320. Reported from firmware  is 320.
  Data in current interval (832 seconds elapsed):
     0 Line Code Violations, 0 Path Code Violations
     0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
     0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
  Total Data (last 55 15 minute intervals):
     0 Line Code Violations, 0 Path Code Violations,
     174 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
     174 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 120 Unavail Secs

===================================================

We have taken debug ccsip all and below is the output,

debug ccsip calls 
SIP Call statistics tracing is enabled

Aug  5 13:28:53.499: //7692/BAFC65BE800F/SIP/Call/sipSPICallInfo: 
The Call Setup Information is:
Call Control Block (CCB) : 0x4B985B90
State of The Call        : STATE_ACTIVE
TCP Sockets Used         : NO
Calling Number           : 8547845654
Called Number            : 7248454145
Source IP Address (Sig  ): 40.40.1.5
Destn SIP Req Addr:Port  : 70.70.1.2:5060
Destn SIP Resp Addr:Port : 70.70.1.2:5060
Destination Name         : abc.sip.com

Aug  5 13:28:53.503: //7692/BAFC65BE800F/SIP/Call/sipSPIMediaCallInfo: 
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 40.40.1.5
Source IP Port    (Media): 18756
Destn  IP Address (Media): 70.70.1.2
Destn  IP Port    (Media): 10872
Orig Destn IP Address:Port (Media): [ - ]:0

Aug  5 13:28:53.507: %ISDN-6-CONNECT: Interface Serial0/0/0:22 is now connected to 7248454145 N/A
Aug  5 13:28:53.575: %ISDN-6-DISCONNECT: Interface Serial0/0/0:22  disconnected from unknown , call lasted 0 seconds
Aug  5 13:28:53.683: //7692/BAFC65BE800F/SIP/Call/sipSPICallInfo: 
The Call Setup Information is:
Call Control Block (CCB) : 0x4B985B90
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 8547845654
Called Number            : 7248454145
Source IP Address (Sig  ): 40.40.1.5
Destn SIP Req Addr:Port  : 70.70.1.2:5060
Destn SIP Resp Addr:Port : 70.70.1.2:5060
Destination Name         : abc.sip.com

Aug  5 13:28:53.683: //7692/BAFC65BE800F/SIP/Call/sipSPIMediaCallInfo: 
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 40.40.1.5
Source IP Port    (Media): 18756
Destn  IP Address (Media): 70.70.1.2
Destn  IP Port    (Media): 10872
Orig Destn IP Address:Port (Media): [ - ]:0

Aug  5 13:28:53.683: //7692/BAFC65BE800F/SIP/Call/sipSPICallInfo: 
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 200


Aug  5 13:31:50.267: //7776/1DBED30688A3/SIP/Call/sipSPICallInfo: 
The Call Setup Information is:
Call Control Block (CCB) : 0x4B98B588
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 1818
Called Number            : 0000211972595243897
Source IP Address (Sig  ): 40.40.1.5
Destn SIP Req Addr:Port  : 50.50.1.1:5075
Destn SIP Resp Addr:Port : 50.50.1.1:5075
Destination Name         : 50.50.1.1

Aug  5 13:31:50.267: //7776/1DBED30688A3/SIP/Call/sipSPIMediaCallInfo: 
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 40.40.1.5
Source IP Port    (Media): 17568
Destn  IP Address (Media): 50.50.1.1
Destn  IP Port    (Media): 5077
Orig Destn IP Address:Port (Media): [ - ]:0

Aug  5 13:31:50.267: //7776/1DBED30688A3/SIP/Call/sipSPICallInfo: 
Disconnect Cause (CC)    : 21
Disconnect Cause (SIP)   : 403

Aug  5 13:38:59.198: //-1/xxxxxxxxxxxx/SIP/Error/sipSPILocateInviteDialogCCB: Could not find ccb for response
Aug  5 13:38:59.198: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
Aug  5 13:38:59.198: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
Aug  5 13:38:59.198: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
Aug  5 13:38:59.198: //8006/2726943C8010/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 40.40.1.5:5060;branch=z9hG4bK24D7180E

From: <sip:8547845654@70.70.1.2>;tag=315EA40-D88

To: <sip:7248454145@abc.sip.com>;tag=10.0.1.10+1+9d416+c9e210eb

Call-ID: 272B76AD-3AAE11E5-88BCBB8F-D9E5C8FF@40.40.1.5

CSeq: 102 INVITE

Timestamp: 1438781937

Server: DC-SIP/2.0

Organization: MetaSwitch

Require: 100rel

Contact: <sip:7248454145@70.70.1.2:5060;transport=udp>

RSeq: 122756028

Content-Length: 0

 


Aug  5 13:38:59.198: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container
Aug  5 13:38:59.198: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentSDP: No SDP found in inbound container
Aug  5 13:38:59.198: //8006/2726943C8010/SIP/Info/ccsip_api_call_alert: SDP Body either absent or ignored in 180 RINGING:- will wait for 200 OK to do negotiation.
Aug  5 13:38:59.198: //8006/2726943C8010/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI headers recvd from app container
Aug  5 13:38:59.198: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No QSIG Body found in inbound container
Aug  5 13:38:59.198: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No RawMsg Body found in inbound container
Aug  5 13:38:59.198: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: No Data to form The Raw Message

Aug  5 13:38:59.198: //8006/2726943C8010/SIP/Info/HandleSIP1xxRinging: ccsip_api_call_alert returned: SIP_SUCCESS
Aug  5 13:38:59.198: //8006/2726943C8010/SIP/State/sipSPIChangeState: 0x4BA2E6A0 : State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)  to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
Aug  5 13:38:59.198: //8006/2726943C8010/SIP/Info/sipSPIPresendProcessing: Presend Processing called for 2 event
Aug  5 13:38:59.198: //8006/2726943C8010/SIP/Info/sipSPI_ipip_GetPassthruCopyListDataFromTdContainer:  Could not get any elements from TD Container
Aug  5 13:38:59.198: //8006/2726943C8010/SIP/Info/sipSPI_ipip_GetPassthruCopyListDataFromTdContainer:  Could not get any elements from TD Container
Aug  5 13:38:59.198: //8006/2726943C8010/SIP/Info/sipSPIGenerateAuthorizationResponse: HA1 is: d9eb1e751b3fdbabb7a4cf0a7ce1880b
Aug  5 13:38:59.198: //8006/2726943C8010/SIP/Info/sipSPISendPrack: Associated container=0x4BEA1B88 to Prack
Aug  5 13:38:59.198: //8006/2726943C8010/SIP/Transport/sipSPISendPrack: Sending PRACK to the transport layer
Aug  5 13:38:59.202: //8006/2726943C8010/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
Aug  5 13:38:59.894: //8006/2726943C8010/SIP/Info/act_recdproc_new_message_response: Transaction Complete. Lock on Facilities released.
Aug  5 13:39:00.634: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [70.70.1.2]:5060, local_address:[ - ]
Aug  5 13:39:00.634: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
Aug  5 13:39:00.634: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
Aug  5 13:39:00.634: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
Aug  5 13:39:00.634: //8006/2726943C8010/SIP/Info/sipSPIMatchRespToReqTran: removing unneeded z9hG4bK24DA3BF from via branch list
Aug  5 13:39:00.634: //8006/2726943C8010/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 200 OK

Via: SIP/2.0/UDP 40.40.1.5:5060;branch=z9hG4bK24DA3BF

From: <sip:8547845654@70.70.1.2>;tag=315EA40-D88

To: <sip:7248454145@abc.sip.com>;tag=10.0.1.10+1+9d416+c9e210eb

Call-ID: 272B76AD-3AAE11E5-88BCBB8F-D9E5C8FF@40.40.1.5

CSeq: 103 PRACK

Server: DC-SIP/2.0

Organization: MetaSwitch

Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event

Supported: resource-priority, 100rel

Content-Length: 0

 


Aug  5 13:39:00.634: //8006/2726943C8010/SIP/Info/sipSPICheckResponseExt: non-INVITE response with no RSEQ - do not disable IS_REL1XX
Aug  5 13:39:00.634: //8006/2726943C8010/SIP/Info/act_recdproc_new_message_response: Transaction Complete. Lock on Facilities released.
Aug  5 13:39:03.583: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x4B98B588) with key=[8004] to table
Aug  5 13:39:03.583: //-1/000000000000/SIP/Info/ccsip_offer_ans_init: 
Aug  5 13:39:03.583: //-1/000000000000/SIP/Info/ccsip_iwf_init:  
Aug  5 13:39:03.583: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init: 
Aug  5 13:39:03.583: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization:  Entry...
Aug  5 13:39:03.583: //-1/000000000000/SIP/Info/ccsipRegisterSetTargetInfo: p2p mode with Registrar Server = dns:abc.sip.com
Aug  5 13:39:03.583: //-1/000000000000/SIP/Info/ccsipRegisterSetTargetInfo: Parsing The Registrar Address
Aug  5 13:39:03.583: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : abc.sip.com target_port : 5060

Aug  5 13:39:03.583: //-1/000000000000/SIP/Info/sipSPIValidateAndCopyOutboundHost: CCSIP: copy target_host to outbound_host
Aug  5 13:39:03.583: //-1/000000000000/SIP/Info/sipSPIOutboundProxyReuse: Do not reuse Outbound Proxy IP adress and Port
Aug  5 13:39:03.583: //-1/000000000000/SIP/State/sipSPIChangeState: 0x4B98B588 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
Aug  5 13:39:03.583: //-1/000000000000/SIP/Info/ccsip_spi_registrar_add_expires_header: Inside ccsip_spi_registrar_add_expires_header for Expires
Aug  5 13:39:03.583: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_OUTBOUND_REGISTER
Aug  5 13:39:03.583: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIIncrementOverloadCount: Local 1 Global 2
Aug  5 13:39:03.583: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 40
Aug  5 13:39:03.583: //-1/000000000000/SIP/Info/act_idle_outgoing_register: In act_idle_outgoing_register

Aug  5 13:39:03.583: //8010/000000000000/SIP/Info/act_idle_outgoing_register:  Send REGISTER to abc.sip.com:5060

Aug  5 13:39:03.583: //8010/000000000000/SIP/Info/sipSPIUaddCcbToUACTable: ****Adding to UAC table.
Aug  5 13:39:03.583: //8010/000000000000/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4B98B588 key=AA598EEA-3A9311E5-8695BB8F-D9E5C8FF
Aug  5 13:39:03.583: //8010/000000000000/SIP/Info/act_idle_outgoing_register: Locally Resolved IP:70.70.1.2:5060
Aug  5 13:39:03.583: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
Aug  5 13:39:03.583: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
Aug  5 13:39:03.583: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 40.40.1.5
Aug  5 13:39:03.583: //8010/000000000000/SIP/Info/sipSPIPresendProcessing: Presend Processing called for 7 event
Aug  5 13:39:03.583: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrieving Data from RCB
Aug  5 13:39:03.583: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrieving Data from RCB
Aug  5 13:39:03.583: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone CDT to SIP default timezone = GMT
Aug  5 13:39:03.587: //8010/000000000000/SIP/Info/sipSPISendRegister: Associated container=0x4BEA3B80 to Register
Aug  5 13:39:03.587: //8010/000000000000/SIP/Transport/sipSPISendRegister: Sending REGISTER to the transport layer
Aug  5 13:39:03.587: //8010/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
Aug  5 13:39:03.587: //8010/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x49DCC8A0, addr=70.70.1.2, port=5060, sentBy_port=0, local_addr=, is_req=1, transport=1, switch=0, callBack=0x41C2EE94
Aug  5 13:39:03.587: //8010/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Aug  5 13:39:03.587: //8010/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Aug  5 13:39:03.587: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:70.70.1.2, rport:5060 with laddr:

Aug  5 13:39:03.587: //8010/000000000000/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x49DCC8A0
Aug  5 13:39:03.587: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x49DCC8A0, addr=70.70.1.2, port=5060, local_addr=, connId=2 for UDP
Aug  5 13:39:03.587: //8010/000000000000/SIP/State/sipSPIChangeState: 0x4B98B588 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
Aug  5 13:39:03.587: //8010/000000000000/SIP/State/sipSPIChangeState: 0x4B98B588 : State change from (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)  to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
Aug  5 13:39:03.587: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
REGISTER sip:abc.sip.com:5060 SIP/2.0

Via: SIP/2.0/UDP 40.40.1.5:5060;branch=z9hG4bK24DBD25

From: <sip:8547845657@abc.sip.com>;tag=31604C0-12EC

To: <sip:8547845657@abc.sip.com>

Date: Wed, 05 Aug 2015 13:39:03 GMT

Call-ID: AA598EEA-3A9311E5-8695BB8F-D9E5C8FF

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1438781943

CSeq: 193 REGISTER

Contact: <sip:8547845657@40.40.1.5:5060>

Expires:  200

Supported: path

Content-Length: 0

 


Aug  5 13:39:04.251: //-1/xxxxxxxxxxxx/SIP/Error/get_content_length: Could not get Content-length
Aug  5 13:39:04.255: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [70.70.1.2]:5060, local_address:[ - ]
Aug  5 13:39:04.255: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone CDT to SIP default timezone = GMT
Aug  5 13:39:04.255: //8010/000000000000/SIP/Info/sipSPISendRegister: Associated container=0x4BEA3B80 to Register
Aug  5 13:39:04.255: //8010/000000000000/SIP/Transport/sipSPISendRegister: Sending REGISTER to the transport layer
Aug  5 13:39:04.255: //8010/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
Aug  5 13:39:04.255: //8010/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x49DCC8A0, addr=70.70.1.2, port=5060, sentBy_port=0, local_addr=, is_req=1, transport=1, switch=0, callBack=0x0
Aug  5 13:39:04.255: //8010/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Aug  5 13:39:04.255: //8010/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Aug  5 13:39:04.255: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:70.70.1.2, rport:5060 with laddr:

Aug  5 13:39:04.255: //8010/000000000000/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x49DCC8A0
Aug  5 13:39:04.255: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x49DCC8A0, addr=70.70.1.2, port=5060, local_addr=, connId=2 for UDP
Aug  5 13:39:04.255: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
Aug  5 13:39:04.255: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
Aug  5 13:39:04.255: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
Aug  5 13:39:04.255: //8010/000000000000/SIP/Info/sipSPIMatchRespToReqTran: removing unneeded z9hG4bK24DBD25 from via branch list
Aug  5 13:39:04.255: //8010/000000000000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 200 OK

Via: SIP/2.0/UDP 40.40.1.5:5060;branch=z9hG4bK24DBD25

From: <sip:8547845657@70.70.1.2>;tag=31604C0-12EC

To: <sip:8547845657@abc.sip.com>;tag=aprqh0gj0l2-db5na8100002c

Call-ID: AA598EEA-3A9311E5-8695BB8F-D9E5C8FF

Timestamp: 1438781943

CSeq: 193 REGISTER

Contact: <sip:8547845657@40.40.1.5:5060>;expires=104

 


Aug  5 13:39:04.255: //8010/000000000000/SIP/Info/sipSPICheckResponseExt: non-INVITE response with no RSEQ - do not disable IS_REL1XX
Aug  5 13:39:04.255: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDecrementOverloadCount: Count:Local 0 Global 1
Aug  5 13:39:04.255: //8010/000000000000/SIP/Info/sipSPIRegPthruProcessResponse: Processing response w/ resp code == 200
Aug  5 13:39:04.255: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetRPCBFromRCB: Retreiving RCB [0x4BD3A5A4] from RPCB [0x0]
Aug  5 13:39:04.255: //8010/000000000000/SIP/Error/sipSPIRegPthruProcessResponse: Error NO RPCB

Aug  5 13:39:04.255: //-1/xxxxxxxxxxxx/SIP/Info/ccsipRegisterStartRCBTimer: Starting timer for pattern 8547845657 for 83 seconds
Aug  5 13:39:04.259: //8010/000000000000/SIP/Info/ccsip_gw_register_process_response: No P-Assoicated-URI present in Register Response
Aug  5 13:39:04.259: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[8004] removed.
Aug  5 13:39:04.259: //8010/000000000000/SIP/Info/sipSPIUdeleteCcbFromUACTable: ****Deleting from UAC table.
Aug  5 13:39:04.259: //8010/000000000000/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x4B98B588 key=AA598EEA-3A9311E5-8695BB8F-D9E5C8FF
Aug  5 13:39:04.259: //8010/000000000000/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
Aug  5 13:39:04.259: //8010/000000000000/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
Aug  5 13:39:04.259: //8010/000000000000/SIP/Info/ccsip_offer_ans_delete: 
Aug  5 13:39:04.259: //8010/000000000000/SIP/Info/ccsip_iwf_delete:  
Aug  5 13:39:04.259: //8010/000000000000/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 4B98B588
Aug  5 13:39:04.259: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[8004]
Aug  5 13:39:04.271: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
REGISTER sip:abc.sip.com:5060 SIP/2.0

Via: SIP/2.0/UDP 40.40.1.5:5060;branch=z9hG4bK24DBD25

From: <sip:8547845657@abc.sip.com>;tag=31604C0-12EC

To: <sip:8547845657@abc.sip.com>

Date: Wed, 05 Aug 2015 13:39:04 GMT

Call-ID: AA598EEA-3A9311E5-8695BB8F-D9E5C8FF

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1438781944

CSeq: 193 REGISTER

Contact: <sip:8547845657@40.40.1.5:5060>

Expires:  200

Supported: path

Content-Length: 0

 


Aug  5 13:39:04.739: //-1/xxxxxxxxxxxx/SIP/Error/get_content_length: Could not get Content-length
Aug  5 13:39:04.739: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [70.70.1.2]:5060, local_address:[ - ]
Aug  5 13:39:04.739: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
Aug  5 13:39:04.739: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
Aug  5 13:39:04.739: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
Aug  5 13:39:04.739: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 200 OK

Via: SIP/2.0/UDP 40.40.1.5:5060;branch=z9hG4bK24DBD25

From: <sip:8547845657@70.70.1.2>;tag=31604C0-12EC

To: <sip:8547845657@abc.sip.com>;tag=aprqh0gj0l2-db5na8100002c

Call-ID: AA598EEA-3A9311E5-8695BB8F-D9E5C8FF

Timestamp: 1438781943

CSeq: 193 REGISTER

Contact: <sip:8547845657@40.40.1.5:5060>;expires=104

 


Aug  5 13:39:04.739: //-1/xxxxxxxxxxxx/SIP/Error/sipSPILocateInviteDialogCCB: Could not find ccb for response
Aug  5 13:39:06.143: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x4B98B588) with key=[8005] to table
Aug  5 13:39:06.143: //-1/000000000000/SIP/Info/ccsip_offer_ans_init: 
Aug  5 13:39:06.143: //-1/000000000000/SIP/Info/ccsip_iwf_init:  
Aug  5 13:39:06.143: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init: 
Aug  5 13:39:06.143: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization:  Entry...
Aug  5 13:39:06.143: //-1/000000000000/SIP/Info/ccsipRegisterSetTargetInfo: p2p mode with Registrar Server = dns:abc.sip.com
Aug  5 13:39:06.143: //-1/000000000000/SIP/Info/ccsipRegisterSetTargetInfo: Parsing The Registrar Address
Aug  5 13:39:06.143: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : abc.sip.com target_port : 5060

Aug  5 13:39:06.143: //-1/000000000000/SIP/Info/sipSPIValidateAndCopyOutboundHost: CCSIP: copy target_host to outbound_host
Aug  5 13:39:06.143: //-1/000000000000/SIP/Info/sipSPIOutboundProxyReuse: Do not reuse Outbound Proxy IP adress and Port
Aug  5 13:39:06.143: //-1/000000000000/SIP/State/sipSPIChangeState: 0x4B98B588 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
Aug  5 13:39:06.143: //-1/000000000000/SIP/Info/ccsip_spi_registrar_add_expires_header: Inside ccsip_spi_registrar_add_expires_header for Expires
Aug  5 13:39:06.143: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_OUTBOUND_REGISTER
Aug  5 13:39:06.143: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIIncrementOverloadCount: Local 1 Global 2
Aug  5 13:39:06.143: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 40
Aug  5 13:39:06.143: //-1/000000000000/SIP/Info/act_idle_outgoing_register: In act_idle_outgoing_register

Aug  5 13:39:06.143: //8011/000000000000/SIP/Info/act_idle_outgoing_register:  Send REGISTER to abc.sip.com:5060

Aug  5 13:39:06.143: //8011/000000000000/SIP/Info/sipSPIUaddCcbToUACTable: ****Adding to UAC table.
Aug  5 13:39:06.143: //8011/000000000000/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4B98B588 key=E522A889-3A9311E5-86A2BB8F-D9E5C8FF
Aug  5 13:39:06.143: //8011/000000000000/SIP/Info/act_idle_outgoing_register: Locally Resolved IP:70.70.1.2:5060
Aug  5 13:39:06.143: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
Aug  5 13:39:06.143: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
Aug  5 13:39:06.143: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 40.40.1.5
Aug  5 13:39:06.143: //8011/000000000000/SIP/Info/sipSPIPresendProcessing: Presend Processing called for 7 event
Aug  5 13:39:06.143: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrieving Data from RCB
Aug  5 13:39:06.143: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrieving Data from RCB
Aug  5 13:39:06.143: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone CDT to SIP default timezone = GMT
Aug  5 13:39:06.147: //8011/000000000000/SIP/Info/sipSPISendRegister: Associated container=0x4BEA75F0 to Register
Aug  5 13:39:06.147: //8011/000000000000/SIP/Transport/sipSPISendRegister: Sending REGISTER to the transport layer
Aug  5 13:39:06.147: //8011/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
Aug  5 13:39:06.147: //8011/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x49DCC8A0, addr=70.70.1.2, port=5060, sentBy_port=0, local_addr=, is_req=1, transport=1, switch=0, callBack=0x41C2EE94
Aug  5 13:39:06.147: //8011/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Aug  5 13:39:06.147: //8011/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Aug  5 13:39:06.147: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:70.70.1.2, rport:5060 with laddr:

Aug  5 13:39:06.147: //8011/000000000000/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x49DCC8A0
Aug  5 13:39:06.147: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x49DCC8A0, addr=70.70.1.2, port=5060, local_addr=, connId=2 for UDP
Aug  5 13:39:06.147: //8011/000000000000/SIP/State/sipSPIChangeState: 0x4B98B588 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
Aug  5 13:39:06.147: //8011/000000000000/SIP/State/sipSPIChangeState: 0x4B98B588 : State change from (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)  to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
Aug  5 13:39:06.147: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent: 
REGISTER sip:abc.sip.com:5060 SIP/2.0

Via: SIP/2.0/UDP 40.40.1.5:5060;branch=z9hG4bK24DC16E1

From: <sip:8547845658@abc.sip.com>;tag=3160EC0-220

To: <sip:8547845658@abc.sip.com>

Date: Wed, 05 Aug 2015 13:39:06 GMT

Call-ID: E522A889-3A9311E5-86A2BB8F-D9E5C8FF

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1438781946

CSeq: 191 REGISTER

Contact: <sip:8547845658@40.40.1.5:5060>

Expires:  200

Supported: path

Content-Length: 0

 


Aug  5 13:39:06.923: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [70.70.1.2]:5060, local_address:[ - ]
Aug  5 13:39:06.927: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone CDT to SIP default timezone = GMT
Aug  5 13:39:06.927: //8011/000000000000/SIP/Info/sipSPISendRegister: Associated container=0x4BEA75F0 to Register
Aug  5 13:39:06.927: //8011/000000000000/SIP/Transport/sipSPISendRegister: Sending REGISTER to the transport layer
Aug  5 13:39:06.927: //8011/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
Aug  5 13:39:06.927: //8011/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x49DCC8A0, addr=70.70.1.2, port=5060, sentBy_port=0, local_addr=, is_req=1, transport=1, switch=0, callBack=0x0
Aug  5 13:39:06.927: //8011/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Aug  5 13:39:06.927: //8011/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Aug  5 13:39:06.927: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:70.70.1.2, rport:5060 with laddr:

Aug  5 13:39:06.927: //8011/000000000000/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x49DCC8A0
Aug  5 13:39:06.927: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x49DCC8A0, addr=70.70.1.2, port=5060, local_addr=, connId=2 for UDP
Aug  5 13:39:06.927: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
Aug  5 13:39:06.927: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
Aug  5 13:39:06.927: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
Aug  5 13:39:06.927: //8011/000000000000/SIP/Info/sipSPIMatchRespToReqTran: removing unneeded z9hG4bK24DC16E1 from via branch list
Aug  5 13:39:06.927: //8011/000000000000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 40.40.1.5:5060;branch=z9hG4bK24DC16E1

From: <sip:8547845658@70.70.1.2>;tag=3160EC0-220

To: <sip:8547845658@abc.sip.com>;tag=10.0.1.10+1+a2105+886a4000

Call-ID: E522A889-3A9311E5-86A2BB8F-D9E5C8FF

Timestamp: 1438781946

CSeq: 191 REGISTER

WWW-Authenticate: Digest realm="abc.sip.com",nonce="5b0025d8e94b",stale=false,algorithm=MD5,qop="auth"

Server: DC-SIP/2.0

Organization: MetaSwitch

Content-Length: 0

 


Aug  5 13:39:06.927: //8011/000000000000/SIP/Info/sipSPICheckResponseExt: non-INVITE response with no RSEQ - do not disable IS_REL1XX
Aug  5 13:39:06.927: //8011/000000000000/SIP/Info/sipSPIGenerateAuthorizationResponse: HA1 is: d9eb1e751b3fdbabb7a4cf0a7ce1880b
Aug  5 13:39:06.927: //8011/000000000000/SIP/Info/sipSPIPresendProcessing: Presend Processing called for 7 event
Aug  5 13:39:06.927: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrieving Data from RCB
Aug  5 13:39:06.931: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrieving Data from RCB
Aug  5 13:39:06.931: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone CDT to SIP default timezone = GMT
Aug  5 13:39:06.931: //8011/000000000000/SIP/Info/sipSPISendRegister: Associated container=0x4BEA7178 to Register
Aug  5 13:39:06.931: //8011/000000000000/SIP/Transport/sipSPISendRegister: Sending REGISTER to the transport layer
Aug  5 13:39:06.931: //8011/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
Aug  5 13:39:06.931: //8011/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x4B4AAC0C, addr=70.70.1.2, port=5060, sentBy_port=0, local_addr=, is_req=1, transport=1, switch=0, callBack=0x41C2EE94
Aug  5 13:39:06.931: //8011/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Aug  5 13:39:06.931: //8011/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Aug  5 13:39:06.931: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:70.70.1.2, rport:5060 with laddr:

Aug  5 13:39:06.931: //8011/000000000000/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x4B4AAC0C
Aug  5 13:39:06.931: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4B4AAC0C, addr=70.70.1.2, port=5060, local_addr=, connId=2 for UDP
Aug  5 13:39:06.931: //8011/000000000000/SIP/State/sipSPIChangeState: 0x4B98B588 : State change from (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)  to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
Aug  5 13:39:06.943: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x4BA28CA8) with key=[8006] to table
Aug  5 13:39:06.943: //-1/000000000000/SIP/Info/ccsip_offer_ans_init: 
Aug  5 13:39:06.947: //-1/000000000000/SIP/Info/ccsip_iwf_init:  
Aug  5 13:39:06.947: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init: 
Aug  5 13:39:06.947: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization:  Entry...
Aug  5 13:39:06.947: //-1/000000000000/SIP/Info/ccsipRegisterSetTargetInfo: p2p mode with Registrar Server = dns:abc.sip.com
Aug  5 13:39:06.947: //-1/000000000000/SIP/Info/ccsipRegisterSetTargetInfo: Parsing The Registrar Address
Aug  5 13:39:06.947: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : abc.sip.com target_port : 5060

Aug  5 13:39:06.947: //-1/000000000000/SIP/Info/sipSPIValidateAndCopyOutboundHost: CCSIP: copy target_host to outbound_host
Aug  5 13:39:06.947: //-1/000000000000/SIP/Info/sipSPIOutboundProxyReuse: Do not reuse Outbound Proxy IP adress and Port
Aug  5 13:39:06.947: //-1/000000000000/SIP/State/sipSPIChangeState: 0x4BA28CA8 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
Aug  5 13:39:06.947: //-1/000000000000/SIP/Info/ccsip_spi_registrar_add_expires_header: Inside ccsip_spi_registrar_add_expires_header for Expires
Aug  5 13:39:06.947: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_OUTBOUND_REGISTER
Aug  5 13:39:06.947: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIIncrementOverloadCount: Local 1 Global 3
Aug  5 13:39:06.947: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 40
Aug  5 13:39:06.947: //-1/000000000000/SIP/Info/act_idle_outgoing_register: In act_idle_outgoing_register

Aug  5 13:39:06.947: //8012/000000000000/SIP/Info/act_idle_outgoing_register:  Send REGISTER to abc.sip.com:5060

Aug  5 13:39:06.947: //8012/000000000000/SIP/Info/sipSPIUaddCcbToUACTable: ****Adding to UAC table.
Aug  5 13:39:06.947: //8012/000000000000/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4BA28CA8 key=1789B3BD-3A9411E5-86B1BB8F-D9E5C8FF
Aug  5 13:39:06.947: //8012/000000000000/SIP/Info/act_idle_outgoing_register: Locally Resolved IP:70.70.1.2:5060
Aug  5 13:39:06.947: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
Aug  5 13:39:06.947: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
Aug  5 13:39:06.947: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 40.40.1.5
Aug  5 13:39:06.947: //8012/000000000000/SIP/Info/sipSPIPresendProcessing: Presend Processing called for 7 event
Aug  5 13:39:06.947: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrieving Data from RCB
Aug  5 13:39:06.947: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIRetrieveOutgoingPassThruData: Retrieving Data from RCB
Aug  5 13:39:06.947: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone CDT to SIP default timezone = GMT
Aug  5 13:39:06.947: //8012/000000000000/SIP/Info/sipSPISendRegister: Associated container=0x4BEA7388 to Register
Aug  5 13:39:06.947: //8012/000000000000/SIP/Transport/sipSPISendRegister: Sending REGISTER to the transport layer
Aug  5 13:39:06.947: //8012/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
Aug  5 13:39:06.947: //8012/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x4B4C7800, addr=70.70.1.2, port=5060, sentBy_port=0, local_addr=, is_req=1, transport=1, switch=0, callBack=0x41C2EE94
Aug  5 13:39:06.947: //8012/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Aug  5 13:39:06.951: //8012/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Aug  5 13:39:06.951: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:70.70.1.2, rport:5060 with laddr:

Aug  5 13:39:06.951: //8012/000000000000/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x4B4C7800
Aug  5 13:39:06.951: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4B4C7800, addr=70.70.1.2, port=5060, local_addr=, connId=2 for UDP
Aug  5 13:39:06.951: //8012/000000000000/SIP/State/sipSPIChangeState: 0x4BA28CA8 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
Aug  5 13:39:06.951: //8012/000000000000/SIP/State/sipSPIChangeState: 0x4BA28CA8 : State change from (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)  to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
Aug  5 13:39:06.951: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
REGISTER sip:abc.sip.com:5060 SIP/2.0

Via: SIP/2.0/UDP 40.40.1.5:5060;branch=z9hG4bK24DC16E1

From: <sip:8547845658@abc.sip.com>;tag=3160EC0-220

To: <sip:8547845658@abc.sip.com>

Date: Wed, 05 Aug 2015 13:39:06 GMT

Call-ID: E522A889-3A9311E5-86A2BB8F-D9E5C8FF

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1438781946

CSeq: 191 REGISTER

Contact: <sip:8547845658@40.40.1.5:5060>

Expires:  200

Supported: path

Content-Length: 0

 


Aug  5 13:39:06.951: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
REGISTER sip:abc.sip.com:5060 SIP/2.0

Via: SIP/2.0/UDP 40.40.1.5:5060;branch=z9hG4bK24DD841

From: <sip:8547845658@abc.sip.com>;tag=3160EC0-220

To: <sip:8547845658@abc.sip.com>

Date: Wed, 05 Aug 2015 13:39:06 GMT

Call-ID: E522A889-3A9311E5-86A2BB8F-D9E5C8FF

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1438781946

CSeq: 192 REGISTER

Contact: <sip:8547845658@40.40.1.5:5060>

Expires: 200

Authorization: Digest username="8547845654",realm="abc.sip.com",uri="sip:abc.sip.com:5060",response="f63b5925fa5232f8f147d80ed03d6a08",nonce="5b0025d8e94b",cnonce="D609F763",qop=auth,algorithm=MD5,nc=00000001

Content-Length: 0

 

Regards,

MUKESH KUMAR | Network Engineer
Spooster IT Services
Computer Networking Solutions
Vivek Batra Tue, 08/25/2015 - 03:33

Can you please disable all debugs except debug ccsip messages... seems couple of SIP messages are missing.

Ayodeji oladipo... Tue, 09/03/2013 - 08:34

Hi,

I have looked at the traces and the problem is from your CUBE. Here is my analysis

1. When ITSP sent the first 180 ringing with SDP, it indicated that it wanted to do early media and requested a PRACK.

"Require: 100rel"

Received:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 21.10.0.3:5060;branch=z9hG4bKBA319D8

From: "ATA187 Core" ;tag=3E21E5CC-34E

To: ;tag=277bus7u-CC-42

Call-ID: AAE9B01E-134F11E3-B2E7A8D4-D9EFC3D@21.10.0.3

CSeq: 101 INVITE

Timestamp: 1378158264

Contact:

Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER

Require: 100rel

RSeq: 1

Content-Length: 228

Content-Type: application/sdp

v=0

o=HuaweiSoftX3000 12465003 12465003 IN IP4 10.57.0.117

s=Sip Call

c=IN IP4 10.57.0.117

t=0 0

m=audio 19728 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=fmtp:18 annexb=no

2. However when CUBE sent a PRACK to the ITSP it included no answer to their offer, ie no SDP.

Sent:

PRACK sip:40042484@10.56.14.18:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 21.10.0.3:5060;branch=z9hG4bKBA4169E

From: "ATA187 Core" ;tag=3E21E5CC-34E

To: ;tag=277bus7u-CC-42

Date: Mon, 02 Sep 2013 21:44:24 GMT

Call-ID: AAE9B01E-134F11E3-B2E7A8D4-D9EFC3D@21.10.0.3

CSeq: 102 PRACK

RAck: 1 101 INVITE

Allow-Events: telephone-event

Max-Forwards: 70

Content-Length: 0

As you can see there is no SDP in the PRACK sent to the ITSP. This is where everything broke. The next few lines, ITSP then sent another 180 ringing without SDP and still requested PRACK, perhaps hoping that CUBE will send offers in its PRACK, but CUBE still ddint send anything...

PRACK is used to establish early media or to cut through audio on the PROGRESS message or ringing in this case. Hence there is no way this can be achieved if CUBE doesnt send SDP in its PRACK message in this scenario.

So we need to find out why CUBE is behaving this way. Please attach a sh run of your gateway.

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

jfernandorosa Tue, 09/03/2013 - 10:02

Hi,

I remove in sip-ua:

disable-early-media 180

The calls in work fine!

Thanks for help!

Regards.

Joao

jfernandorosa Tue, 09/03/2013 - 11:20

Hi,

After this change, my fax services don´t work, I call to number receive a  ring, receive a signal and return a fast busy.

I attached logs.

Joao

Ayodeji oladipo... Tue, 09/03/2013 - 11:35

I can see that the call starts as g729 and changed to G711 when a fax tone was detected at the far end..

What is the ip address of the device your fax is connected to? What is the region setting between your fax device and the gateway?

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

jfernandorosa Tue, 09/03/2013 - 11:50

IP Endpoint ATA187 is: 21.10.1.50

Region between Fax and Gateway is G711

.

Thanks!

Joao

Ayodeji oladipo... Tue, 09/03/2013 - 12:05

You need to reconfigure the dial-peers for your fax devices. The voice call setup is sent to a xcoder..

From the logs, RTP stream is sent to ip address 21.10.0.2.

Received:

ACK sip:297832422526@21.10.0.3:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 21.10.0.7:5060;branch=z9hG4bK153f6bc5cebe

From: "ATA187 Core" ;tag=24382~fb89236f-816b-47f5-8c94-b8d3c388dd7c-64665213

To: ;tag=4272C5B4-D17

Date: Tue, 03 Sep 2013 17:53:30 GMT

Call-ID: bcc58e80-2261221a-c15-7000a15@21.10.0.7

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: presence, kpml

Content-Type: application/sdp

Content-Length: 251

v=0

o=CiscoSystemsCCM-SIP 24382 1 IN IP4 21.10.0.7

s=SIP Call

c=IN IP4 21.10.0.2

That suggest that the call is routed to a transcoder or MTP device...can you confirm what device is this.

If the region between your ATA and CUBE is G711, then you need to have the inbound dial-peer from CUCM advertise G711 and your outbound dial-peer to your ITSP using G711 for fax calls only, all other calls can use G729

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepare

jfernandorosa Tue, 09/03/2013 - 12:12

Hi,

21.10.0.2 is a CUBE with MTP and transcoder.

I forced call to use G711 in dial-peer (in and out), but not successfully, attached logs with G711.

Thanks.

Joao

Ayodeji oladipo... Tue, 09/03/2013 - 12:28

I can see the call using G711 however rtp is still sent to c=IN IP4 21.10.0.6.

For your fax to work media/rtp stream has to be sent directly to the ATA.

Looking at your config, I can see that this device is a MTP device. So there is a DTMF mistmact hence calls are terminated on the MTP..

Try and add this to your inbound voip dial-peer

dial-peer voice 2 voip

description # Call Leg VOIP #

incoming called-number .

dtmf-relay rtp-nte digit-drop sip-kpml

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

jfernandorosa Tue, 09/03/2013 - 12:35

21.10.0.6 is a CUCM.

I can´t insert the RTP comands, in inbound dial-peer is not exist:

ra076963.igs.cref(config-dial-peer)#dtmf-relay rtp-nte digit-drop ?

  cisco-rtp          Cisco Proprietary RTP

  h245-alphanumeric  DTMF Relay via H245 Alphanumeric IE

  h245-signal        DTMF Relay via H245 Signal IE

 

I insert:

dial-peer voice 2 voip

description # Call Leg VOIP #

incoming called-number .

dtmf-relay rtp-nte digit-drop

Not working.

Thanks!

Joao

Ayodeji oladipo... Tue, 09/03/2013 - 12:40

Dont you have a sip trunk between CUCM and CUBE? If you do then your dial-peer 2 should have session protocol sipv2 on it. You need the dial-peer to be enabled for sip to add that command

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

jfernandorosa Tue, 09/03/2013 - 12:56

Yes, I have the SIP Trunk between CUCM and CUBE.

I change the dial-peer to:

dial-peer voice 2 voip

description # Call Leg VOIP #

session protocol sipv2

incoming called-number .

voice-class codec 1

dtmf-relay rtp-nte digit-drop sip-kpml

But not working.

Attached logs.

Thanks!

Joao.

Ayodeji oladipo... Tue, 09/03/2013 - 13:08

The call is still sent to MTP not the fax device. What do you have configured on your sip trunk dtmf method? Is it rfc2833 or no preference?

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Ayodeji oladipo... Tue, 09/03/2013 - 13:14

ok..Can you go to the ATA187 config on cucm and check if you have the option of "RFC2833 enable"..If its there please selct the tick box and reset the phone then test again

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

jfernandorosa Tue, 09/03/2013 - 13:21

In ATA187 don´t have this option.

Have (Require DTMF Reception) uncheck.

Thanks!

jfernandorosa Tue, 09/03/2013 - 13:39

Hi,

I attached new log fax_problem_log4.txt.zip

IP ATA187 is 21.10.0.14

In log IP the ATA is ok in RTP.

v=0

o=CiscoSystemsSIP-GW-UserAgent 4424 2824 IN IP4 21.10.0.3

s=SIP Call

c=IN IP4 21.10.0.14  -->IP ATA187

t=0 0

m=audio 16386 RTP/AVP 8 101

c=IN IP4 21.10.0.14

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

Thanks!

Joao

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