Solved: My UC dont make voip call

lima.pedro · ‎10-03-2013

Good morning sirs,

I finish config my uc but isnt make call to voip server, just to pstn. I put the preference as 1 to sip and 10 to pstn.

I dial and wait and wait.... and go by pstn. On my sip server my uc show me ok in registar and dont have no one try to calls going to server.

When I made a dial plan test its show me go by sip and after pstn. So my dial plan is ok, i think.

I believe its is firewall problem. Have a command to debug a sip voip call ? to see where its stop?

And could I test a acl? On Asa I have this command. :S

Any help is welcome! thanks so much.

Alexander Maroukian · ‎10-07-2013

Hello Pedro,

You may try to add first choice to voice class codec to be g729r8 because in the other working one you are using g729r8 codec (this is the default in the dialpeer when nothing else is set) as far as I can see.

HTH,

Alex

*Please rate helpful posts

View solution in original post

Alexander Maroukian · ‎10-03-2013

Hello Pedro,

Are the incoming calls working?

Best regards,

Alex

lima.pedro · ‎10-04-2013

Yeah, normally work. I think was the firewall, but I remove all rules to test and isnt is. Now I belive is the dial plan. I remove all dial plans and put just one to work and the calss leave.

But with extremelly delay to start call. Anyone know why this problem is now?

thanks too much.

lima.pedro · ‎10-04-2013

I repair to work but I exclude many lines. I dont know what can be. Just to compair, I will put the original conf with the repair conf.

The original conf is the first.

!

stcapp ccm-group 1

stcapp

!

isdn switch-type primary-net5

!

trunk group ALL_FXO

max-retry 5

voice-class cause-code 1

hunt-scheme longest-idle

!

trunk group ALL_T1E1

hunt-scheme longest-idle

translation-profile outgoing PROFILE_ALL_T1E1

!

voice call send-alert

voice rtp send-recv

!

voice service voip

ip address trusted list

ipv4 0.0.0.0 0.0.0.0

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol none

no fax-relay sg3-to-g3

sip

registrar server expires max 3600 min 3600

localhost dns:restrict

outbound-proxy dns:restrict

no update-callerid

sip-profiles 1000

!

voice class codec 1

codec preference 1 g711ulaw

!

voice class sip-profiles 1000

request ANY sdp-header Connection-Info remove

response ANY sdp-header Connection-Info remove

!

voice class cause-code 1

no-circuit

!

voice register global

mode cme

source-address 10.1.1.1 port 5060

load 9971 sip9971.9-2-2

load 9951 sip9951.9-2-2

load 8961 sip8961.9-2-2

timezone 17

time-format 24

date-format D/M/Y

!

voice source-group CCA_SIP_SOURCE_GROUP_CUE_CME

access-list 2

translation-profile incoming SIP_Incoming

!

voice source-group CCA_SIP_SOURCE_GROUP_EXTERNAL

access-list 3

!

voice translation-rule 4

rule 15 /^....$/ /1234xxx/

!

voice translation-rule 6

rule 1 /xx00/ /xx00/

rule 2 /xx01/ /xx01/

rule 3 /xx02/ /xx02/

rule 4 /xx03/ /xx03/

rule 5 /xx04/ /xx04/

rule 6 /xx05/ /xx05/

rule 7 /xx06/ /xx06/

rule 8 /xx07/ /xx07/

rule 9 /xx08/ /xx08/

rule 10 /xx09/ /xx09/

!

voice translation-rule 7

rule 1 /xx10/ /xx10/

rule 2 /xx11/ /xx11/

rule 3 /xx12/ /xx12/

rule 4 /xx13/ /xx13/

rule 5 /xx14/ /xx14/

rule 6 /xx15/ /xx15/

rule 7 /xx16/ /xx16/

rule 8 /xx17/ /xx17/

rule 9 /xx18/ /xx18/

rule 10 /xx19/ /xx19/

!

voice translation-rule 8

rule 1 /xx20/ /xx20/

rule 2 /xx21/ /xx21/

rule 3 /xx22/ /xx22/

rule 4 /xx23/ /xx23/

rule 5 /xx24/ /xx24/

rule 6 /xx25/ /xx25/

rule 7 /xx26/ /xx26/

rule 8 /xx27/ /xx27/

rule 9 /xxxx/ /xxxx/

rule 10 /xx29/ /xx29/

!

voice translation-rule 9

rule 1 /xx30/ /xx30/

rule 2 /xx31/ /xx31/

rule 3 /xx32/ /xx32/

rule 4 /xx33/ /xx33/

rule 5 /xx34/ /xx34/

rule 6 /xx35/ /xx35/

rule 7 /xx36/ /xx36/

rule 8 /xx37/ /xx37/

rule 9 /xx38/ /xx38/

rule 10 /xx39/ /xx39/

!

voice translation-rule 10

rule 1 /xx40/ /xx40/

rule 2 /xx41/ /xx41/

rule 3 /xx42/ /xx42/

rule 4 /xx43/ /xx43/

rule 5 /xx44/ /xx44/

rule 6 /xx45/ /xx45/

rule 7 /xx46/ /xx46/

rule 8 /xx47/ /xx47/

rule 9 /xx48/ /xx48/

rule 10 /xx49/ /xx49/

!

voice translation-rule 11

rule 1 /xx50/ /xx50/

rule 2 /xx51/ /xx51/

rule 3 /xx52/ /xx52/

rule 4 /xx53/ /xx53/

rule 5 /xx54/ /xx54/

rule 6 /xx55/ /xx55/

rule 7 /xx56/ /xx56/

rule 8 /xx57/ /xx57/

rule 9 /xx58/ /xx58/

rule 10 /xx59/ /xx59/

!

voice translation-rule 12

rule 1 /xx60/ /xx60/

rule 2 /xx61/ /xx61/

rule 3 /xx62/ /xx62/

rule 4 /xx63/ /xx63/

rule 5 /xx64/ /xx64/

rule 6 /xx65/ /xx65/

rule 7 /xx66/ /xx66/

rule 8 /xx67/ /xx67/

rule 9 /xx68/ /xx68/

rule 10 /xx69/ /xx69/

!

voice translation-rule 13

rule 1 /xx70/ /xx70/

rule 2 /xx71/ /xx71/

rule 3 /xx72/ /xx72/

rule 4 /xx73/ /xx73/

rule 5 /xx74/ /xx74/

rule 6 /xx75/ /xx75/

rule 7 /xx76/ /xx76/

rule 8 /xx77/ /xx77/

rule 9 /xx78/ /xx78/

rule 10 /xx79/ /xx79/

!

voice translation-rule 14

rule 1 /xx80/ /xx80/

rule 2 /xx81/ /xx81/

rule 3 /xx82/ /xx82/

rule 4 /xx83/ /xx83/

rule 5 /xx84/ /xx84/

rule 6 /xx85/ /xx85/

rule 7 /xx86/ /xx86/

rule 8 /xx87/ /xx87/

rule 9 /xx88/ /xx88/

rule 10 /xx89/ /xx89/

!

voice translation-rule 15

rule 1 /xx90/ /xx90/

rule 2 /xx91/ /xx91/

rule 3 /xx92/ /xx92/

rule 4 /xx93/ /xx93/

rule 5 /xx94/ /xx94/

rule 6 /xx95/ /xx95/

rule 7 /xx96/ /xx96/

rule 8 /xx97/ /xx97/

rule 9 /xx98/ /xx98/

rule 10 /xx99/ /xx99/

!

voice translation-rule 410

rule 1 /^0$.*$/ /\1/

rule 15 /^....$/ /134711/

!

voice translation-rule 411

rule 1 /^0$.*$/ /ABCD0\1/

!

voice translation-rule 412

rule 1 /^ABCD$.*$/ /\1/

!

voice translation-rule 422

rule 15 /^ABCD$.*$/ /\1/

!

voice translation-rule 1000

rule 1 /.*/ //

!

voice translation-rule 1111

rule 15 /^....$/ /134711/

!

voice translation-rule 1112

rule 1 /^0/ //

!

voice translation-rule 2000

rule 1 /xx99/ /xx99/

!

voice translation-rule 2002

rule 1 /^6/ //

!

voice translation-rule 2222

!

voice translation-profile CALLER_ID_TRANSLATION_PROFILE

translate calling 1111

!

voice translation-profile CallBlocking

translate called 2222

!

voice translation-profile DDR_Called_10

translate calling 3265

translate called 10

!

voice translation-profile DDR_Called_11

translate calling 3265

translate called 11

!

voice translation-profile DDR_Called_12

translate calling 3265

translate called 12

!

voice translation-profile DDR_Called_13

translate calling 3265

translate called 13

!

voice translation-profile DDR_Called_14

translate calling 3265

translate called 14

!

voice translation-profile DDR_Called_15

translate calling 3265

translate called 15

!

voice translation-profile DDR_Called_6

translate calling 3265

translate called 6

!

voice translation-profile DDR_Called_7

translate calling 3265

translate called 7

!

voice translation-profile DDR_Called_8

translate calling 3265

translate called 8

!

voice translation-profile DDR_Called_9

translate calling 3265

translate called 9

!

voice translation-profile OUTGOING_TRANSLATION_PROFILE

translate called 1112

!

voice translation-profile PROFILE_ALL_T1E1

translate calling 4

!

voice translation-profile PSTN_CallForwarding

translate redirect-target 410

translate redirect-called 410

!

voice translation-profile PSTN_Outgoing

translate calling 1111

translate called 1112

translate redirect-target 410

translate redirect-called 410

!

voice translation-profile SIP_Incoming

translate called 411

!

voice translation-profile SIP_Passthrough

translate called 412

!

voice translation-profile SIP_Passthrough_CallBlocking

translate called 422

!

voice translation-profile VM_Profile

translate called 2000

!

voice translation-profile XFER_TO_VM_PROFILE

translate redirect-called 2002

!

voice translation-profile nondialable

translate called 1000

!

voice-card 0

dspfarm

dsp services dspfarm

!

fax interface-type fax-mail

!

license udi pid UC560-T1E1-K9 sn restrict

archive

log config

logging enable

logging size 600

hidekeys

username Restrict privilege 15 secret 4 Restrict

!

controller E1 0/3/0

pri-group timeslots 1-31

!

ip tftp source-interface Vlan90

!

interface Serial0/3/0:15

no ip address

encapsulation hdlc

isdn switch-type primary-net5

isdn incoming-voice voice

isdn supp-service name calling

isdn sending-complete

isdn outgoing ie redirecting-number

trunk-group ISDN1 1

no cdp enable

!

sccp local Vlan90

sccp ccm 10.1.1.1 identifier 1 version 4.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate profile 2 register mtpRestrictd6620

associate profile 1 register confprof1

!

dspfarm profile 2 transcode

description CCA transcoding for SIP Trunk Restrict

codec g711ulaw

codec g729ar8

codec g729abr8

maximum sessions 10

associate application SCCP

!

dspfarm profile 1 conference

description DO NOT MODIFY, active CCA conference profile - CCA2.0 codec711

codec g711ulaw

codec g711alaw

maximum sessions 4

conference-join custom-cptone CCAjointone

conference-leave custom-cptone CCAleavetone

associate application SCCP

!

dial-peer cor custom

name internal

name local

name local-plus

name international

name national

name national-plus

name emergency

name toll-free

name PSTN-fax

!

dial-peer cor list call-internal

member internal

!

dial-peer cor list call-local

member local

!

dial-peer cor list call-local-plus

member local-plus

!

dial-peer cor list call-national

member national

!

dial-peer cor list call-national-plus

member national-plus

!

dial-peer cor list call-international

member international

!

dial-peer cor list call-emergency

member emergency

!

dial-peer cor list call-toll-free

member toll-free

!

dial-peer cor list user-internal

member internal

member emergency

!

dial-peer cor list user-local

member internal

member local

member emergency

member toll-free

!

dial-peer cor list user-local-plus

member internal

member local

member local-plus

member emergency

member toll-free

!

dial-peer cor list user-national

member internal

member local

member local-plus

member national

member emergency

member toll-free

!

dial-peer cor list user-national-plus

member internal

member local

member local-plus

member national

member national-plus

member emergency

member toll-free

!

dial-peer cor list user-international

member internal

member local

member local-plus

member international

member national

member national-plus

member emergency

member toll-free

!

dial-peer cor list call-fax

member PSTN-fax

!

dial-peer voice 6 pots

description "catch all dial peer for BRI/PRI"

translation-profile incoming nondialable

incoming called-number .%

direct-inward-dial

!

dial-peer voice 50 pots

description ** incoming dial peer **

incoming called-number ^AAAA$

port 0/1/0

!

dial-peer voice 51 pots

description ** incoming dial peer **

incoming called-number ^AAAA$

port 0/1/1

!

dial-peer voice 52 pots

description ** incoming dial peer **

incoming called-number ^AAAA$

port 0/1/2

!

dial-peer voice 53 pots

description ** incoming dial peer **

incoming called-number ^AAAA$

port 0/1/3

!

dial-peer voice 54 pots

description ** FXO pots dial-peer **

destination-pattern A0

port 0/1/0

no sip-register

!

dial-peer voice 55 pots

description ** FXO pots dial-peer **

destination-pattern A1

port 0/1/1

no sip-register

!

dial-peer voice 56 pots

description ** FXO pots dial-peer **

destination-pattern A2

port 0/1/2

no sip-register

!

dial-peer voice 57 pots

description ** FXO pots dial-peer **

destination-pattern A3

port 0/1/3

no sip-register

!

dial-peer voice 2000 voip

description ** cue voicemail pilot number **

translation-profile outgoing XFER_TO_VM_PROFILE

destination-pattern xx99

b2bua

session protocol sipv2

session target ipv4:10.1.10.1

voice-class sip outbound-proxy ipv4:10.1.10.1

dtmf-relay sip-notify

codec g711ulaw

no vad

!

dial-peer voice 2002 voip

description ** cue voicemail PSTN number **

translation-profile outgoing VM_Profile

destination-pattern xx99$

b2bua

session protocol sipv2

session target ipv4:10.1.10.1

voice-class sip outbound-proxy ipv4:10.1.10.1

dtmf-relay sip-notify

codec g711ulaw

no vad

!

dial-peer voice 1112 pots

destination-pattern xx23

port 0/0/2

!

dial-peer voice 1113 pots

destination-pattern xx90

port 0/0/3

!

dial-peer voice 58 pots

trunkgroup ALL_T1E1

corlist outgoing call-local

description **CCA*Brazil*local**

translation-profile outgoing OUTGOING_TRANSLATION_PROFILE

preference 3

destination-pattern 0[2-4]T

forward-digits all

no sip-register

!

dial-peer voice 59 pots

trunkgroup ALL_FXO

corlist outgoing call-local

description **CCA*Brazil*local**

translation-profile outgoing OUTGOING_TRANSLATION_PROFILE

preference 5

destination-pattern 0[2-4]T

forward-digits all

no sip-register

!

dial-peer voice 10xx voip

!

dial-peer voice 1020 voip

corlist outgoing call-international

description **CCA*Brazil*Celular teste**

translation-profile outgoing PSTN_Outgoing

preference 1

destination-pattern 09xx8....

session protocol sipv2

session target sip-server

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

!

dial-peer voice 60 pots

trunkgroup ALL_T1E1

corlist outgoing call-local

description **CCA*Brazil*Celluar 8 digitos**

translation-profile outgoing OUTGOING_TRANSLATION_PROFILE

preference 3

destination-pattern 0[5-9].......

forward-digits all

no sip-register

!

dial-peer voice 61 pots

trunkgroup ALL_FXO

corlist outgoing call-local

description **CCA*Brazil*Celluar 8 digitos**

translation-profile outgoing OUTGOING_TRANSLATION_PROFILE

preference 5

destination-pattern 0[5-9].......

forward-digits all

no sip-register

!

dial-peer voice 1002 voip

description ***TESTE SIP TRUNK ***

translation-profile outgoing Outgoing-VOIP

preference 1

destination-pattern 0.T

session protocol sipv2

session target sip-server

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

ip qos dscp cs5 media

ip qos dscp cs4 signaling

!

no dial-peer outbound status-check pots

sip-ua

credentials username 1user pass realm

authentication username 1user pass realm

no remote-party-id

retry invite 2

retry register 1

timers connect 100

registrar dns:dns expires 900

sip-server dns:dns

!

I think is a problem with dial peer, but I dont know where exactly is. When I remove the calls go normally.

Please, anyone can help me?

Alexander Maroukian · ‎10-07-2013

Hello Pedro,

You may try to add first choice to voice class codec to be g729r8 because in the other working one you are using g729r8 codec (this is the default in the dialpeer when nothing else is set) as far as I can see.

HTH,

Alex

*Please rate helpful posts

lima.pedro · ‎10-08-2013

Hello Alexander, thanks so much, its work now and the priority of trunk on dial peer need changes too

But you help me a lot, thanks againa man!