09-12-2012 09:17 AM - edited 03-14-2019 10:30 AM
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.
09-13-2012 02:15 AM
Hi,
You can check the document :
section :
Dialer_Detail Table Database Fields and Descriptions --- on Page 190 , if that what is you looking for ..
Thanks A lot ,
Ahmed Salah
09-13-2012 03:31 AM
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.
09-13-2012 06:54 AM
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
09-13-2012 07:04 AM
Hi, sure, post your gateway config. G.
09-17-2012 09:10 AM
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
09-17-2012 11:21 PM
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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