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Luis Yrigoyen
Enthusiast

Outbound Option -- SIP dialer callresultdetail field in Dialer_Detail table

Hi,

We're planning on converting from SCCP dialer to SIP dialer.  I did the transition in our lab environment but I'm getting a bunch of errors when it comes to dialing--it seems that it works when it wants to.

I've noticed that in the Dialer_Detail table, the CallResultDetail colunm now has result codes but I can't find their definitions any where.

I'm getting codes 404 and 10503. 

I figured the 404 is the SIP code for busy but what's 10503?

Any advise on using SIP dialer with PRI trunks?

thanks

We're running UCCE 8.5 with CVP 8.0 and CM 8.6.

6 REPLIES 6
Ahmed Khalefa
Beginner

Hi,

You can check the document :

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/outbound_option/outboundoption8_5/installation/guide/icm85otb.pdf

section :

Dialer_Detail Table Database Fields and Descriptions  --- on Page 190 , if that what is you looking for ..

Thanks A lot ,

Ahmed Salah

Ahmed, bravo, that was a really quick answer, but unfortunately, I am not able to see the codes 404 and 10503 anywheere in the document you provided. Were you able to see them? If yes, where?

Luis,

According to the Databas Schema Guide (available here:

http://www.cisco.com/en/US/products/sw/custcosw/ps1844/prod_technical_reference_list.html) column CallResultDetail is "Reserved for future use" in 8.5, even in 9.0. "Reserved for future use" means you leave it alone, you don't try to figure out its meaning as it might change in the future. Simple.

Can you please tell me more about the errors you're getting in your lab environment?

"it seems that it works when it wants to" - come on, seriously. Try to understand it before you go and say such things. If you suspect a configruation error, again, try to understand it and fix it. If you suspect a bug, file a bug report. But if it really works when it wants to, then congratulations, you found AI hidden in Outbound Option.

I have some experience with SIP dialling in OO, with PRI trunks, too. So the only thing you need to do is ask real questions. 

G.

Gergely,

     Correct, I read the guides and I noticed the "Reserved for future use" but what cought my attention was that previous to using the SIP dialer, the callresultdetail filed hadn't been populated.  I figured maybe it had been reserved for the SIP dialer........anyway, I was just trying to make sense of the codes to try to troubleshoot the problem I'm having.

In the lab I have UCCE 8.5, CVP 8.0, CM 8.6, GW 2811 with 15.1T.

In the GW I have 2 FXO ports attached to analog lines and also a SIP trunk (provided by our service provider).

Aside from the SIP dialer, everything works, Inbound calls to CM ext., inbound calls to CVP, outbound calls from CM via FXOs, outbound calls from CM via SIP trunk.  Outbound calls with the SCCP dialer (all modes predictive, preview, etc.)

The SIP dialer was working fine (testing in predictive mode) and all of a sudden, without changing anything, I started getting SIP error code 487 and 200 (seen in the GW ccsip debug) and the dialer reports a CallResult 6 (no dial tone).

SIP code 487 means "Request has terminated by bye or cancel" and code 200 means "the request was successful" but in either case the call doesn't connect. 

Can I send you my GW config so you can maybe look at it and tell me if Im missing something?  It's very simple config.

I'm thinking it has to do with some delay with the FXO ports???? I'm planning on testing it with a PRI to see what results I get.

thanks

Hi, sure, post your gateway config. G.

Thanks so much, below is the config:

version 15.1

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname LAB-RS-ING_VXML-GW

!

boot-start-marker

boot-end-marker

!

enable secret 5 $1$3To0$gEXE6LwLcPYMB5Jt3033J0

!

no aaa new-model

no network-clock-participate slot 1

!

voice-card 0

!

voice-card 1

dspfarm

!

ip source-route

!

!

ip cef

!

!

ip domain name sscincorporated.com

ip host mediaserver 10.x.x.83

no ipv6 cef

multilink bundle-name authenticated

!

!

!

!

!

trunk group FXO-Ports

hunt-scheme sequential

!

!

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

signaling forward none

h323

sip     

  bind control source-interface FastEthernet0/0

  bind media source-interface FastEthernet0/0

  header-passing

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 5 g729r8

!

voice class h323 1

  h225 timeout tcp establish 3

!

!

!

!

voice translation-rule 99

rule 1 /^4/ /334/

!

!

voice translation-profile PROFILE_TO_CVP

translate called 99

!

!

http client cache memory pool 15000

http client cache memory file 500

ivr prompt memory 15000

ivr record memory session 20000

!

application

service new-call flash:bootstrap.vxml

  paramspace english index 0

  paramspace english language en

  paramspace english location flash

  paramspace english prefix en

!

service ringtone flash:ringtone.tcl

  paramspace english index 0

  paramspace english language en

  paramspace english location flash

  paramspace english prefix en

!

service cvperror flash:cvperror.tcl

  paramspace english index 0

  paramspace english language en

  paramspace english location flash

  paramspace english prefix en

!

service survive flash:survivability.tcl

  param keepalive survive

  paramspace english language en

  paramspace english index 0

  paramspace english location flash:

  param alert-timeout 8

  param after-hours-agent0 7672

  paramspace english prefix en

  paramspace callfeature med-inact-det enable

  param setup-timeout 7

!

service bootstrap flash:bootstrap.tcl

  paramspace english index 0

  paramspace english language en

  paramspace english location flash

  paramspace english prefix en

!

service handoff flash:handoff.tcl

  paramspace english language en

  paramspace english index 0

  paramspace english location flash

  paramspace english prefix en

!

!

!

!

license udi pid CISCO2811 sn FTX1020A2X3

archive

log config

  hidekeys

!

!

!

!

!

!

!

interface Loopback0

ip address 172.20.1.1 255.255.255.255

!

interface FastEthernet0/0

ip address 10.x.x.254 255.255.255.0

duplex auto

speed auto

!

!

router eigrp 15

network 10.0.0.0

auto-summary

!

ip forward-protocol nd

!

!

no ip http server

ip rtcp report interval 2000

ip route 0.0.0.0 0.0.0.0 10.x.x.254

!

!

control-plane

!

!

voice-port 0/0/0

trunk-group FXO-Ports 1

supervisory disconnect dualtone mid-call

output attenuation -3

no echo-cancel enable

no non-linear

no vad

playout-delay maximum 250

playout-delay nominal 200

playout-delay minimum high

playout-delay mode fixed

timeouts call-disconnect 5

timeouts wait-release 5

connection plar 4930

description DID 305-826-4930

!

voice-port 0/0/1

trunk-group FXO-Ports 2

supervisory disconnect dualtone mid-call

output attenuation -3

no echo-cancel enable

no non-linear

no vad

playout-delay maximum 250

playout-delay nominal 200

playout-delay minimum high

playout-delay mode fixed

timeouts call-disconnect 5

timeouts wait-release 5

connection plar 4198

description DID 305-826-4198

!

voice-port 0/0/2

!

voice-port 0/0/3

!

!

!

!

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 2 voip

incoming called-number .

voice-class codec 1

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 987654 voip

description Fixes Programming Workaround by Blocking 987654

translation-profile incoming block

incoming called-number 987654

!

dial-peer voice 9191 voip

description SIP Ringtone Dial-Peer

service ringtone

incoming called-number 91919191

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 9292 voip

description SIP Error Dial-Peer

service cvperror

incoming called-number 92929292

voice-class sip rel1xx disable

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 99 voip

description For Incoming Leg (Type 10 label and Correlation ID)

service bootstrap

incoming called-number 3399T

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 4930 voip

description 305-826-4930

translation-profile outgoing PROFILE_TO_CVP

destination-pattern 4930

session protocol sipv2

session target ipv4:10.x.x.83

voice-class codec 1

LAB-RS-ING_VXML-GW#sho run | beg dial-peer

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 2 voip

incoming called-number .

voice-class codec 1

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 987654 voip

description Fixes Programming Workaround by Blocking 987654

translation-profile incoming block

incoming called-number 987654

!

dial-peer voice 9191 voip

description SIP Ringtone Dial-Peer

service ringtone

incoming called-number 91919191

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 9292 voip

description SIP Error Dial-Peer

service cvperror

incoming called-number 92929292

voice-class sip rel1xx disable

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 99 voip

description For Incoming Leg (Type 10 label and Correlation ID)

service bootstrap

incoming called-number 3399T

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 4930 voip

description 305-826-4930

translation-profile outgoing PROFILE_TO_CVP

destination-pattern 4930

session protocol sipv2

session target ipv4:10.x.x.83

voice-class codec 1

dtmf-relay rtp-nte

no vad

!

dial-peer voice 50001001 pots

incoming called-number 4930

trunk-group-label source FXO-Ports

direct-inward-dial

!        

dial-peer voice 786 voip

description ** Inbound Calls via SIP Trunk ***

destination-pattern 786.......

session protocol sipv2

session target ipv4:10.x.x.81

no voice-class sip outbound-proxy  

dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 4198 voip

description 305-826-4198

destination-pattern 4198

session target ipv4:10.x.x.81

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 301 pots

trunkgroup FXO-Ports

description ** MIA Local Calls **

destination-pattern 305[2-9]......

progress_ind alert enable 8

progress_ind progress enable 8

direct-inward-dial

prefix 305

!

!

!

call-manager-fallback

secondary-dialtone 9

max-conferences 2 gain -6

transfer-system full-consult

limit-dn 7936 1

limit-dn 7940 10

limit-dn 7941 10

limit-dn 7942 10

timeouts interdigit 6

Hi,

try the following lines in dial peer 786:

voice-class sip rel1xx supported "100rel"

no voice-class sip reset timer expires 183

Good luck.

G.

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