Re: about a call back in UC520

borhen bouallagui · ‎02-24-2012

Hello

need help about a call back in UC500.

from Internal to External

Ip Phone --> UC520 -->GSM Gateway(2N voiceblue lite) OK!!

from external

my GSM number--->GSM Gateway (2N voiceblue lite) -->UC520 {cue auto attendant number } -->ip phone OK!!!

when i make call trought GSM Gateway (2N voiceblue lite) from may cisco ip phone all is good and perfect.

but after a few minute i recive a call in my Cisco ip phone and in my GSM number .

the probleme that the uc500 make a call betwen my cisco ip phone and my GSM number automaticllly after a 10 min of my call.

---> Gsm gateway informs me that the 910 {cue auto attendant} it was he who made the call between the cisco ip phone ad the GSM.

so i maked a test in other UC520 without {cue auto attendant number configuref } and work fine without call bak so the probleme in the

{cue auto attendant}

how can fix this problem i need to use cue auto attendant in my environnement. ???????

------------------------------------------------------------------

dial-peer voice 2001 voip

description ** cue auto attendant number **

translation-profile outgoing PSTN_CallForwarding

destination-pattern 910

b2bua

session protocol sipv2

session target ipv4:10.1.10.1

voice-class sip outbound-proxy ipv4:10.1.10.1

dtmf-relay rtp-nte.

codec g711ulaw

no vad

-------------------------------------------------------------

dial-peer voice 90 voip --- to GSM Gateway

corlist outgoing call-national

destination-pattern [2549].......

session protocol sipv2

session target ipv4:10.1.1.10

codec g711alaw

no vad

rowseyba1 · ‎02-24-2012

Borhen,

I think your the problem you are having is that a phantom call comes back to you after a few minutes of you hanging up a call? Is that correct?

borhen bouallagui · ‎02-24-2012

yes this is my problem.

rowseyba1 · ‎02-24-2012

I have a couple of questions.

What kind of phone service is used to connect you to the PSTN? PRI? POTS? SIP Trunk?

Have you run any debug's to see where the call is originating? Is it the system or does it come from your service provider?

What is the version of OS you are running? Do a show version in CLI.

Knowing this information would help to track down your information

borhen bouallagui · ‎02-25-2012

first i use a sip trunk beteew uc520 and mobile gateway.

the call is from systeme and exactelly from 910 the extension for cue auto attendant number.

version of OS is:

Cisco IOS Software, UC500 Software (UC500-ADVIPSERVICESK9-M), Version 15.1(2)T4, RELEASE SOFTWARE (fc1)

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

Compiled Tue 12-Jul-11 22:01 by prod_rel_team

ROM: System Bootstrap, Version 12.4(11r)XW3, RELEASE SOFTWARE (fc1)

UC_GSS uptime is 16 hours, 52 minutes

System returned to ROM by power-on

System restarted at 16:48:34 CET Fri Feb 24 2012

System image file is "flash:uc500-advipservicesk9-mz.151-2.T4"

Last reload type: Normal Reload

Note:

this probleme come if activation of cue auto attendant. if i disable this option there are no probleme.

borhen bouallagui · ‎02-28-2012

I steel need help

rowseyba1 · ‎02-28-2012

I understand that the call is coming from the AA, but where is it originating? Have you checked with your SIP provider to see if there are duplicate calls being originated for each call coming in?

The problem sounds like a call loop of some kind. If it's something easily repeatable, set up a syslog server and capture the following debugs.

debug ccsip messages

debug voice ccapi inout

borhen bouallagui · ‎03-02-2012

Hi

in attached the file CSV of debug.

pls i need help the client steel wait for explanation

rowseyba1 · ‎03-02-2012

So here's what I see.

I see that the original call comes from here:

910@192.168.10.1 and 10.100

Then a secondary call comes from 910@10.1.10.1

There seems to be some double configuration here. Can you send a copy of the show run with usernames/passwords edited out?

Also, can someone else confirm what I'm seeing?

borhen bouallagui · ‎03-05-2012

hi

this is the config

thx

rowseyba1 · ‎03-05-2012

Borhen,

I though you said you were using a SIP trunk? The configuration that you sent is POTS lines. There isn't a SIP trunk set up on this box. I've never set up a UC520 this way, but what "may" be happening is that two of your POTS lines are being engaged when a call comes in.

Does anyone else have more experience with this type of setup?

borhen bouallagui · ‎03-05-2012

NOT SIP TRUNK . I USE H323

dial-peer voice 90 voip

corlist outgoing call-national

destination-pattern 0[2549].......

session protocol sipv2

session target ipv4:192.168.10.100

codec g711alaw

no vad

!

dial-peer voice 91 pots

corlist outgoing call-national

description ** FXO pots dial-peer **

preference 1

destination-pattern 0[2549].......

port 0/3/3

!

dial-peer voice 92 pots

corlist outgoing call-national

description ** FXO pots dial-peer **

preference 2

destination-pattern 0[2549].......

port 0/1/0

!

dial-peer voice 93 pots

corlist outgoing call-national

description ** FXO pots dial-peer **

preference 3

destination-pattern 0[2549].......

port 0/1/1

!

dial-peer voice 94 pots

corlist outgoing call-national

description ** FXO pots dial-peer **

preference 4

destination-pattern 0[2549].......

port 0/1/2

!

dial-peer voice 95 pots

corlist outgoing call-national

description ** FXO pots dial-peer **

preference 5

destination-pattern 0[2549].......

port 0/1/3

!

dial-peer voice 96 pots

corlist outgoing call-national

description ** FXO pots dial-peer **

preference 6

destination-pattern 0[2549].......

port 0/2/0

!

dial-peer voice 97 pots

corlist outgoing call-national

description ** FXO pots dial-peer **

preference 7

destination-pattern 0[2549].......

port 0/2/1

!

dial-peer voice 98 pots

corlist outgoing call-national

description ** FXO pots dial-peer **

preference 8

destination-pattern 0[2549].......

port 0/2/2

THE FIRST dial-peer route the call to GSM gatway, if busy or noan the call will be redirected to FXO port {8 port fxo} with preference.

rowseyba1 · ‎03-05-2012

borhen bouallagui wrote:

first i use a sip trunk beteew uc520 and mobile gateway.

the call is from systeme and exactelly from 910 the extension for cue auto attendant number.

version of OS is:

Cisco IOS Software, UC500 Software (UC500-ADVIPSERVICESK9-M), Version 15.1(2)T4, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2011 by Cisco Systems, Inc.
Compiled Tue 12-Jul-11 22:01 by prod_rel_team
ROM: System Bootstrap, Version 12.4(11r)XW3, RELEASE SOFTWARE (fc1)
UC_GSS uptime is 16 hours, 52 minutes
System returned to ROM by power-on
System restarted at 16:48:34 CET Fri Feb 24 2012
System image file is "flash:uc500-advipservicesk9-mz.151-2.T4"
Last reload type: Normal Reload

Note:

this probleme come if activation of cue auto attendant. if i disable this option there are no probleme.

Apologies, but you said you used a SIP trunk.

Your setup is outside the realm of my experience. I think you will need to either open a case with TAC or someone else with more experience with your type of setup will need to help.