Unable to transfer call coming from PSTN

[techguy]

Hi,

I have h323 voice gateway cisco 2611XM with NM-2V card (means VIC-2FXS and VIC-2FXO). Its connected with CUCM 7.1. Internal extension can transfer the calls but calls coming PSTN cant tranfer the calls to internal extension. Its disconnecting immeditely.

Is there any codec problem or do i need any dsp resourse ?. 

[acampbell]

Hi,

Can you test if this same IP phone can put external callers ON HOLD for say 5 seconds then RESUME.

If this does not work then the issue is MTP related.

When a Xfer is set up it uses HOLD.

Let us know the results.

HTH

Alex

[techguy]

i did what you said. I can hold the call from that phone set then i did unhold. It works i mean to say call wasnt disconnected.

[acampbell]

Hi,

Thank you for the update.

This then means the issue looks to be CODEC related.

Can we see the config from your gateway.

Regards

Alex

[techguy]

hostname XXXXXX

!

boot-start-marker

boot system flash

boot-end-marker

!

enable secret 5 XXXXXXXX

!

no aaa new-model

clock timezone PCTime 4

no network-clock-participate slot 1

no network-clock-participate wic 0

ip cef

!

!

!

!

!

!

!

!

!

trunk group  FXO

!

!

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

h323

sip

!

!

!

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

codec preference 3 g729r8

!

!

!

voice class h323 1

h225 timeout tcp establish 3

!

!

!

!

!

!

!

!

!

!

application

  global

  service alternate DEFAULT

!

!

!

interface Loopback105

ip address 10.1.209.193 255.255.255.192

h323-gateway voip interface

h323-gateway voip bind srcaddr 10.1.209.193

!

!

ip http server

no ip http secure-server

!

!

voice-port 1/0/0

description FXS port

signal groundStart

echo-cancel coverage 32

bearer-cap Speech

caller-id enable

!

voice-port 1/0/1

description FXS port

signal groundStart

!

voice-port 1/1/0

description FXO port

trunk-group FXO

input gain -3

output attenuation -1

echo-cancel coverage 32

no comfort-noise

timing hookflash-out 50

connection plar opx 2700

impedance 900c

music-threshold -45

bearer-cap Speech

!

voice-port 1/1/1

description FXO port

trunk-group FXO

input gain -3

output attenuation -1

echo-cancel coverage 32

no comfort-noise

timing hookflash-out 50

connection plar opx 2700

impedance 900c

music-threshold -45

bearer-cap Speech

!

!

!

sccp local Loopback105

sccp ccm 10.50.2.3 identifier 1 version 4.1

sccp ccm 10.50.2.4 identifier 2 version 4.1

sccp

!

sccp ccm group 1

associate ccm 2 priority 2

associate ccm 1 priority 1

associate profile 3 register ABUDHABI_MTP

associate profile 2 register ABUDHABI_XCODE

associate profile 1 register ABUDHABI_CONF

!

dspfarm profile 2 transcode

associate application SCCP

shutdown

!

dspfarm profile 1 conference

associate application SCCP

shutdown

!

dspfarm profile 3 mtp

codec g711ulaw

maximum sessions software 10

associate application SCCP

!

!

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 2 pots

trunkgroup FXO

tone ringback alert-no-PI

description Mobile

destination-pattern 905........$

fax rate disable

prefix 05

!

dial-peer voice 3 pots

tone ringback alert-no-PI

description National

destination-pattern 90[234678].......$

fax rate disable

port 1/1/0

prefix 0

!

dial-peer voice 4 pots

description Emergency

destination-pattern 9[19]..$

fax rate disable

port 1/1/0

forward-digits 3

!

dial-peer voice 7000 voip

preference 1

destination-pattern 99..

voice-class codec 1

voice-class h323 1

session target ipv4:10.50.2.3

dtmf-relay h245-alphanumeric

req-qos guaranteed-delay audio

acc-qos guaranteed-delay audio

fax-relay ecm disable

fax rate 14400

fax nsf 000000

fax protocol pass-through g711ulaw

no vad

!

dial-peer voice 7001 voip

preference 2

destination-pattern 270.

voice-class codec 1

voice-class h323 1

session target ipv4:10.50.2.3

dtmf-relay h245-alphanumeric

req-qos guaranteed-delay audio

acc-qos guaranteed-delay audio

fax-relay ecm disable

fax rate 14400

fax nsf 000000

fax protocol pass-through g711ulaw

no vad

!

dial-peer voice 10 pots

trunkgroup FXO

tone ringback alert-no-PI

description International

destination-pattern 900.T

fax rate disable

prefix 00

!

dial-peer voice 301 voip

description CUE

destination-pattern 9913

session protocol sipv2

session target ipv4:10.1.209.62

dtmf-relay sip-notify

codec g711ulaw

no vad

!

!

!

!

telephony-service

max-ephones 10

max-dn 10

ip source-address 10.1.209.193 port 2000

time-zone 21

time-format 24

date-format dd-mm-yy

max-conferences 8 gain -6

transfer-system full-consult

!

!

line con 0

exec-timeout 0 0

speed 115200

line aux 0

line vty 0

password XXXXXX

login

line vty 1 4

password XXXXXX

login

!

scheduler max-task-time 5000

scheduler allocate 4000 1000

ntp clock-period 17210598

ntp server 192.117.105.69

ntp server 219.87.217.84

ntp server 152.118.24.8

ntp server 163.25.109.16

!

end

[acampbell]

Hi,

I see that the H323 Ip address is 10.1.209.193

Can all the phones see this address.

Also I see your codec class is this order

!

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

codec preference 3 g729r8

!

but your MTP is:-

!

dspfarm profile 3 mtp

codec g711ulaw

maximum sessions software 10

associate application SCCP

!

I would change the CODEC class to be :-

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

!

HTH

Alex

[techguy]

I changed the codec order but it did not work...

[acampbell]

Hi,

What about IP connectivity from the gateway on Ip address 10.1.209.193 to all the phones

Regards

Alex

[techguy]

IP Phones can ping 10.1.209.193. This ip address is loopback address.

Let me tell you something that operator can transfer on the extensions which are not in that office. (does not matter that on which route partition they are.) Does it seem routing problem ?

[jbankstonfla]

I just fixed a nearly identical problem in my lab.

FXO 1/0/0 - pstn line 1

FXO 1/0/1 - pstn line 2

FXS 1/1/0 - fax

FXS 1/1/1 - analog phone

my problem was that without a dsp farm resources, all inbound calls were forcing G.729. I went under my voip peer of the H.323 gateway and forced G.711 inbound to CUCM. I also found a code issue when I try to forward a call to the analog phone, both fax and analog phone are trying to answer. When I disconnected the fax, the analog phone works perfectly, so my issue has to do the FXS ports in a NM-2V in an H.323 2610XM router.

After forcing G.711, everything cleared up including access to Unity for voicemail, inbound calls from pstn conf to IP phones, etc. all of it.

Keep in mind that I don't have a dsp farm, just using the media resources from CUCM.

-Jeff

[techguy]

i am not not using FXS port for any purpose. I am using only IP Phones which are registered on CUCM. When a calls comes gateway (via FXO port) where i have connection plar of reception extension which is registered on cucm. It goes through VoIP dial -peer.

Anyways when calls comes direct to reception extension then he cant transfer that line to any user.

[acampbell]

Hi

I see your router is an old 2611

Have you checked for IOS bugs

May be you need to upgrade

Show version in posting.

Regards

Alex

[SajithThrimavithana]

This worked for us, after forcing g.711 everything worked nicely. Thank you !