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Cisco Sip Dialer (8.5) problems with customer call

I have SIP Dialer 8.5 is conected directly to a voice gateway Cisco 2851 with IOS c2800nm-entservicesk9-mz.151-3.T4.bin, regisered 2 phones with number 1010 and 1020 at CUCM 8.6.

All services PG and Dialer in active state.

Login in CTIOS Desktop (phone number: 1010, id agent: 7500). For test I make a call from Dialer to number 1020.

Agent reserved system but call fail.

Log file from gateway:

dialer.png

10.3.3.3 - Dialer

10.3.3.11 - Gateway

10.3.3.6 - CUCM

448 - dialer port

1010 - agent phone

1020 - customer phone

Dec 16 13:30:02.269: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:1020@10.3.3.11 SIP/2.0

Via: SIP/2.0/UDP 10.3.3.3:58800;branch=z9hG4bK-d8754z-89061269042a8a62-1---d8754z-;rport

Max-Forwards: 70

Require: 100rel

Contact: <sip:448@10.3.3.3:58800>

To: <sip:1020@10.3.3.11>

From: <sip:448@10.3.3.3>;tag=fd38e32f

Call-ID: ea7a8534-9e02137e-44052e46-8e39203c

CSeq: 1 INVITE

Session-Expires: 1800

Min-SE: 90

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, NOTIFY, PRACK, REFER, NOTIFY, OPTIONS

Content-Type: Multipart/mixed;boundary=uniqueBoundary

Supported: timer, resource-priority, replaces

User-Agent: Cisco-SIPDialer/UCCE8.0

Content-Length: 530

Remote-Party-ID: <sip:@10.3.3.11>;party=calling;screen=no;privacy=off

--uniqueBoundary

Content-Type: application/sdp

Content-Disposition: session;handling=required

v=0

o=CiscoSystemsSIP-GW-UserAgent 2884 2524 IN IP4 172.19.155.41

s=SIP Call

c=IN IP4 0.0.0.0

t=0 0

m=audio 19994 RTP/AVP 0

a=inactive

--uniqueBoundary

Content-Type: application/x-cisco-cpa

Content-Disposition: signal;handling=optional

Events=FT,Asm,AsmT,Sit,Piano

CPAMinSilencePeriod=608

CPAAnalysisPeriod=2500

CPAMaxTimeAnalysis=3000

CPAMaxTermToneAnalysis=30000

CPAMinValidSpeechTime=112

--uniqueBoundary--

Source Filename: ccapi_16_12_dialer1

*Dec 16 13:30:02.277: //18653/006583A78908/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.3.3.3:58800;branch=z9hG4bK-d8754z-89061269042a8a62-1---d8754z-;rport

From: <sip:448@10.3.3.3>;tag=fd38e32f

To: <sip:1020@10.3.3.11>

Date: Mon, 16 Dec 2013 13:30:02 GMT

Call-ID: ea7a8534-9e02137e-44052e46-8e39203c

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Source Filename: ccapi_16_12_dialer1

Dec 16 13:30:02.289: //18654/006583A78908/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:1020@10.3.3.6:5060 SIP/2.0

Via: SIP/2.0/UDP 10.3.3.11:5060;branch=z9hG4bK62588

From: <sip:10.3.3.11>;tag=5B51AFC-1423

To: <sip:1020@10.3.3.6>

Date: Mon, 16 Dec 2013 13:30:02 GMT

Call-ID: 67F4E0-658D11E3-890EF0A5-EBD428B4@10.3.3.11

Supported: timer,resource-priority,replaces,sdp-anat

Require: 100rel

Min-SE:  1800

Cisco-Guid: 0006652839-1703743971-2299064485-3956549812

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

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

CSeq: 101 INVITE

Timestamp: 1387200602

Contact: <sip:10.3.3.11:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Session-Expires:  1800

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 211

v=0

o=CiscoSystemsSIP-GW-UserAgent 310 2097 IN IP4 10.3.3.11

s=SIP Call

c=IN IP4 10.3.3.11

t=0 0

m=audio 18030 RTP/AVP 0 19

c=IN IP4 10.3.3.11

a=rtpmap:0 PCMU/8000

a=rtpmap:19 CN/8000

a=ptime:20

Source Filename: ccapi_16_12_dialer1

Dec 16 13:30:02.313: //18654/006583A78908/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 420 Bad Extension

Via: SIP/2.0/UDP 10.3.3.11:5060;branch=z9hG4bK62588

From: <sip:10.3.3.11>;tag=5B51AFC-1423

To: <sip:1020@10.3.3.6>;tag=2062172177

Date: Mon, 16 Dec 2013 13:39:16 GMT

Call-ID: 67F4E0-658D11E3-890EF0A5-EBD428B4@10.3.3.11

CSeq: 101 INVITE

Allow-Events: presence

Unsupported: 100rel

Content-Length: 0

Source Filename: ccapi_16_12_dialer1

Dec 16 13:30:02.317: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:1020@10.3.3.6:5060 SIP/2.0

Via: SIP/2.0/UDP 10.3.3.11:5060;branch=z9hG4bK62588

From: <sip:10.3.3.11>;tag=5B51AFC-1423

To: <sip:1020@10.3.3.6>;tag=2062172177

Date: Mon, 16 Dec 2013 13:30:02 GMT

Call-ID: 67F4E0-658D11E3-890EF0A5-EBD428B4@10.3.3.11

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Source Filename: ccapi_16_12_dialer1

Dec 16 13:30:02.317: //18653/006583A78908/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 500 Internal Server Error

Via: SIP/2.0/UDP 10.3.3.3:58800;branch=z9hG4bK-d8754z-89061269042a8a62-1---d8754z-;rport

From: <sip:448@10.3.3.3>;tag=fd38e32f

To: <sip:1020@10.3.3.11>;tag=5B51B1C-2647

Date: Mon, 16 Dec 2013 13:30:02 GMT

Call-ID: ea7a8534-9e02137e-44052e46-8e39203c

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=127

Content-Length: 0

Source Filename: ccapi_16_12_dialer1

17 Replies 17

Gergely Szabo
VIP Alumni
VIP Alumni

Hi,

alright, did you check the usual suspects, for instance, is there a SIP trunk between the GW and CUCM. Also, can you post the outbound SIP dial-peer from the GW to CUCM. Should be something like this:

dial-peer voice 2001 voip

  description SIP REFER -> CUCM

  destination-pattern .T

  session protocol sipv2

  session target ipv4:10.20.30.40 !CUCM address

  voice-class sip rel1xx supported "100rel"

  no voice-class sip reset timer expires 183

  dtmf-relay sip-notify

  codec g711ulaw

  no vad

G.

Hi,

voice service voip

ip address trusted list

  ipv4 10.3.3.0 255.255.255.0

allow-connections sip to sip

signaling forward none

sip

  bind control source-interface Loopback22

  bind media source-interface Loopback22

  rel1xx require "100rel"

dial-peer voice 51 voip

session protocol sipv2

incoming called-number .T

voice-class sip rel1xx require "100rel"

codec g711ulaw

dial-peer voice 52 voip

session protocol sipv2

destination-pattern 1...

voice-class sip rel1xx require "100rel"

session target ipv4:10.3.3.6 !CUCM address

codec g711ulaw

Hi, can you just try replacing the voice-class sip line in the outbound dial-peer with voice-class sip rel1xx supported "100rel".

G.

I change voice-class sip rel1xx supported "100rel" for dial-peer's,

but the problem persists

Gergely Szabo
VIP Alumni
VIP Alumni

Hi, can you please rewrite the destionation-pattern expression in the outbound dial-peer to .T - if possible, just for testing.
Also can you confirm the IOS version is 15.x? It won't work with older versions.
G.


Sent from Cisco Technical Support Android App

Нi, I am rewrite dial-peer, and now dial-peer look like:

dial-peer voice 52 voip

session protocol sipv2

destination-pattern .T

voice-class sip rel1xx supported "100rel"

session target ipv4:10.3.3.6 !CUCM address

codec g711ulaw

#sh ver

Cisco IOS Software, 2800 Software (C2800NM-ENTSERVICESK9-M), Version       15.1(3)T4, RELEASE SOFTWARE (fc1)

Technical Support: http://www.cisco.com/techsupport

Copyright (c) 1986-2012 by Cisco Systems, Inc.

ROM: System Bootstrap, Version 12.3(8r)T7, RELEASE SOFTWARE (fc1)

Okay, I would like you to set up SIP tracing in CUCM. Do you know how to do that?

Thanks.

G.

Trace file from CUCM in attachment (enable all detail trace)

Hi,

this

15:10:58.696 |//SIP/SIPHandler/ccbId=0/scbId=0/getRel1xxType: No matching SIP trunk found in hash table, returning rel1xx disabled|1,100,230,1.50^10.3.3.11^*

15:10:58.696 |//SIP/SIPHandler/ccbId=0/scbId=0/sipSPIGetCallExtensionSupported: SIPRel1xxEnabledServiceParamSetting=0 , ccb->pld.outboundRel1xx=1|1,100,230,1.50^10.3.3.11^*

15:10:58.696 |//SIP/Stack/Error/0xf3a3340/Bad Extension - Require header processing failed.|1,100,230,1.50^10.3.3.11^*

is a bit suspicious.

Is there a SIP trunk configured in CUCM, pointing towards the GW?

Also, can you please try removing the rel1xx require "100rel" command from the voice service voip > sip section?

Also, is there a CVP involved in here?

G.

Hi,

New configuration gateway, new debug call (agent login on phone 1010, call customer phone 1020, 1030, 0635878424)

I dont have CVP, I use IPIVR

On CUCM I have trunk with destination 10.3.3.11, specific security profile and normalisatin script according Outbound Option Guide

for Cisco Unified Contact Center Enterprise and Hosted 8.5(1)

And I add in sip profile some change (problem not resolve):

Gateway:

voice service voip

ip address trusted list

  ipv4 10.202.0.0 255.255.0.0

allow-connections sip to sip

signaling forward none

sip

  bind control source-interface Loopback202

  bind media source-interface Loopback202

dial-peer voice 51 voip

session protocol sipv2

incoming called-number .T

voice-class sip rel1xx supported "100rel"

codec g711ulaw

!

dial-peer voice 52 voip

description to_CCM

destination-pattern .T

session protocol sipv2

session target ipv4:10.3.3.6

voice-class sip rel1xx supported "100rel"

no voice-class sip reset timer expires 183

dtmf-relay sip-notify

codec g711ulaw

no vad

Log call on number 1020(cucm) - not call, 1030(cucm) - not call, (0635878424(pri) - when dialer call on mobile number I recive call, but call not transfer to agent - file CUCM in attachtment log_cucm.txt

OK, let's take a look at it from a different angle.

What does the agent see? Can you give me a screenshot of it?

G.

Reservation agent:

when I recive call on mobile phone, string reservation disappears

Oops.

You should use the other CTI Toolkit Desktop, the one with the Outbound Option buttons (Accept, Reject, etc).

If you chose the default installation options, it's

"C:\Program Files\Cisco Systems\CTIOS Client\CTIOS Toolkit\Win32 CIL\Samples\CTI Toolkit Outbound Desktop\CTIOSOutOptSSoftphone.exe"

Can you also confirm that the dialing mode is Preview or Preview-Direct?

Thanks.

G.

In CTI Toolkit Outbound Desktop same situation, only buttons not active when reservation occure...

Dialing mode is Predictive

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