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Community Member

CANT COMPLETE A CALL FROM LAN TO PSTN (VoIP-CALLMANAGER-ROUTER)

PETER, thanks for your reply.

I will try to improve my english. (My native language is the spanish)

I was working with G711 because bandwitdh was enougth BUT Now I need to reduce bandwitdh usage for incoming and outgoing calls.

Peter I have noticed something: It could be a CODEC's setting because

everything happens when I change the REGION's Codec in the Call manager from G711 to G723.

In my LAN Ip phones can comunicate when I am using G723 but when I try to make an outgoing call It doesnt work.

When I say that I can not complete a call I mean:

Internal calls.

-I can dial a internal phone in my LAN.

-I can hear the calling's ring.

-The another end can hung up the phone.

-I can talk without problems.

Outgoing calls.

-I can dial a outside phone (for example: a cellular phone, my home's phone (PSTN)).

-I can hear the calling's ring.

-BUT after a second the call's ring stops and a Busy tone sounds; then The another end can not hung up the phone.And there wasnt any error messages or entries in the log or displayed on IP phones.z

-For example, in my cell phone I can see incoming call but automatically after a second it is considered as a missing call. I have not enough time to hung up the phone.

Incoming calls.

-I can dial the HEADER CORPORATE NUMBER and also the IP phone's extension from my cell phone.

-I can hear the calling's ring.

-In the Ip phone I can see the incoming call but when I hung up it I olny hear a busy tone. At the same time in my cell phone the time counter go on but the is no sound.

I THINK,there is something missing in the Router.

It is enough changing the codec prefrerence like this?

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

The dial peer configurated for INCOMING calls Is:

dial-peer voice 2 pots

description LLAMADAS CELULARES

preference 2

destination-pattern 9.......

progress_ind alert enable 8

progress_ind progress enable 8

progress_ind connect enable 8

no digit-strip

direct-inward-dial

port 1/0:15

forward-digits all

!

dial-peer voice 20 pots

description LLAMADAS CELULARES

preference 1

destination-pattern 9.......

progress_ind alert enable 8

progress_ind progress enable 8

progress_ind connect enable 8

no digit-strip

direct-inward-dial

port 1/1:15

forward-digits all

!

the dial peer configurated for OUTGOING calls Is:

dial-peer voice 100 voip

preference 1

destination-pattern [2-8]...

progress_ind setup enable 3

voice-class h323 1

session target ipv4:172.20.64.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 101 voip

preference 2

destination-pattern [2-8]...

progress_ind setup enable 3

voice-class h323 1

session target ipv4:172.20.64.4

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 102 voip

preference 1

destination-pattern 1[1-3]..

progress_ind setup enable 3

voice-class h323 1

session target ipv4:172.20.64.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 103 voip

preference 2

destination-pattern 1[1-3]..

progress_ind setup enable 3

voice-class h323 1

session target ipv4:172.20.64.4

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 104 voip

preference 1

destination-pattern 1[5-9]..

progress_ind setup enable 3

voice-class h323 1

session target ipv4:172.20.64.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 105 voip

preference 2

destination-pattern 1[5-9]..

progress_ind setup enable 3

voice-class h323 1

session target ipv4:172.20.64.4

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

Today I am going to apply some debug commands and I will post it.

Thanks in advance!

WALTER MUNGUIA

SAN MARCOS UNIVERSITY

CCNA

6 REPLIES
Community Member

Re: CANT COMPLETE A CALL FROM LAN TO PSTN (VoIP-CALLMANAGER-ROUT

thi is sh ver:

********************************** GW_UNMSM#sh ver

Cisco IOS Software, 3600 Software (C3640-IS-M), Version 12.3(11)T5, RELEASE SOFTWARE (fc1)

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

Copyright (c) 1986-2005 by Cisco Systems, Inc.

Compiled Sat 02-Apr-05 12:23 by yiyan

ROM: System Bootstrap, Version 11.1(20)AA2, EARLY DEPLOYMENT RELEASE SOFTWARE (fc1)

GW_UNMSM uptime is 3 days, 17 hours, 36 minutes

System returned to ROM by reload

System image file is "flash:c3640-is-mz.123-11.T5.bin"

Cisco 3640 (R4700) processor (revision 0x00) with 89088K/9216K bytes of memory.

Processor board ID 26856110

R4700 CPU at 100MHz, Implementation 33, Rev 1.0

2 FastEthernet interfaces

62 Serial interfaces

2 Channelized E1/PRI ports

DRAM configuration is 64 bits wide with parity disabled.

125K bytes of NVRAM.

32768K bytes of processor board System flash (Read/Write)

Configuration register is 0x2102

***********************************

Community Member

Re: CANT COMPLETE A CALL FROM LAN TO PSTN (VoIP-CALLMANAGER-ROUT

Hello everybody,

There is someting strage happening in my VoIP-LAN network.

I can not complete a call from my IP phone to the PSTN when I set my 3640 Router.

This is scenario:

-----<3640 ROUTER>------

We are working with one REGION configured in a Call Manager V3.3

When the the only one REGION is set to use the G711 we can complete the call easyly but when we change the Region codec from G711 to G729 we can complete internal call (in my LAN) but we can not complete call to outside the LAN anymore.

Bronze

Re: CANT COMPLETE A CALL FROM LAN TO PSTN (VoIP-CALLMANAGER-ROUT

hi,

you should configure "voice class codec 1" on the VOIP dial-peers. the codec selection should be dynamic in this case.

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

dial-peer voice XXXX voip

preference 1

destination-pattern XXXX

progress_ind setup enable 3

NO codec g711ulaw

VOICE-CLASS CODEC 1

session target ipv4:172.20.64.5

dtmf-relay h245-alphanumeric

INCOMING CALLED NUMBER .

no vad

!

greetings

mehmet

Hall of Fame Super Silver

Re: CANT COMPLETE A CALL FROM LAN TO PSTN (VoIP-CALLMANAGER-ROUT

The reason is because you hard coded your dial-peers to g711, here is what you need to do:

either change the codec to g729, or create a codec class and apply to every voip dialpeer, such as:

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

dial-peer voice 102 voip

preference 1

destination-pattern 1[1-3]..

progress_ind setup enable 3

voice-class codec 1

voice-class h323 1

session target ipv4:172.20.64.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

Please rate all useful posts!

Chris

Community Member

Re: CANT COMPLETE A CALL FROM LAN TO PSTN (VoIP-CALLMANAGER-ROUT

thanx Mehmet

thanx Chris

you both gave me the KEY for the huge problem!

I applied "voice-class codec 1" in my voip dial peers and now I can complete G729 and G711 calls depending the REGION(with a particular codec) they are in the Call Manager.

In this moment I am applying the G729 codec into 10 REGIONs. Now My CAMPUS will use G729 codec when I make a outgoing call to my 10 Remote Sites.

Now i will have a improved bandwitdh comsuption.

THANKS AGAIN.

SOME QUESTIONS.

cderen (CHRIS):

In the configuration you gave me:

dial-peer voice 102 voip

preference 1

destination-pattern 1[1-3]..

progress_ind setup enable 3

voice-class codec 1

voice-class h323 1

session target ipv4:172.20.64.5

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

I think there is no reason to use "codec g711ulaw " bacause it is defined in my voice-class codec 1. Is it true?

**********************************

msolak(MEHMET)

You add a diferent command line in the config you gave me:

dial-peer voice XXXX voip

preference 1

destination-pattern XXXX

progress_ind setup enable 3

NO codec g711ulaw

VOICE-CLASS CODEC 1

session target ipv4:172.20.64.5

dtmf-relay h245-alphanumeric

INCOMING CALLED NUMBER .

no vad

!

What else can this line do? "INCOMING CALLED NUMBER . "

now I have everything working. Do I have to add it?

THANX FRIENDS

WALTER MUNGUIA

SAN MARCOS UNIVERSITY

LIMA-PERU

Hall of Fame Super Silver

Re: CANT COMPLETE A CALL FROM LAN TO PSTN (VoIP-CALLMANAGER-ROUT

You are correct I forgot to remove the "codec g711ulaw" line.

Please rate all helpful posts!

Chris

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