SIP provider with Call Manager 3.2, 3725 GW

[seth.wilson]

I have a customer with Call Manager 3.2 and a 3725 GW. The customer wants to start using a SIP provider. The network is currently set up H.323 entirely. Do I need IP-to-IP multiservice IOS version in order to make this work? Thanks.

[gogasca]

Yes, you need IP-to-IP GW feature since CCM 3.2 doesnt support SIP.

[seth.wilson]

I got outbound calls working with 12.4.10 IP VOICE. File name is c3725-ipvoicek9-mz.124-10. Relevant config looks like this:

voice service voip

allow-connections h323 to sip

allow-connections sip to h323

h323

sip

bind control source-interface Multilink1

bind media source-interface Multilink1

!

translation-rule 1

Rule 0 ^71 1

!

dial-peer voice 200 voip

service session

destination-pattern 7T

translate-outgoing called 1

session protocol sipv2

session target ipv4:172.20.100.6

incoming called-number

dtmf-relay rtp-nte h245-alphanumeric

codec g711ulaw

!

dial-peer voice 210 voip

destination-pattern

progress_ind setup enable 3

session target ipv4:10.10.253.238

dtmf-relay h245-alphanumeric

codec g711ulaw

!

sip-ua

sip-server ipv4:172.20.100.6

!

sh call active voice br shows the following:

Telephony call-legs: 0

SIP call-legs: 1

H323 call-legs: 1

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 2

1287 : 90 7568810ms.1 +-1 pid:100 Answer connecting

dur 00:00:00 tx:0/0 rx:0/0

IP 172.20.100.6:10056 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711u

law

media inactive detected:n media contrl rcvd:n/a timestamp:n/a

1287 : 93 7591320ms.1 +-1 pid:110 Originate connecting

dur 00:00:00 tx:0/0 rx:0/0

IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-iet

f

media inactive detected:n media contrl rcvd:n/a timestamp:n/a

Telephony call-legs: 0

SIP call-legs: 1

H323 call-legs: 1

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 2

Incoming calls however do not work. Here are the symptoms. When an outside caller dials the number, the outside caller hears ringback indefinitely, regardless if the user picks up the handset or not. When the SCCP IP phone user picks up, user hears dead air for a few seconds and then fast busy.

Codec on the H.323 leg shows G.729r8. Very strange since I specified in the dial peer G.711ulaw and both the gateway and the phone are in the same device pool and region and are configured to use G.711 within the region.

I assume that the call is failing due to a codec mismatch...

Help...

[thisisshanky]

Can you do a codec g729r8 on both legs and see if the call works ? (Just to test)

[seth.wilson]

The provider claims that they can only support G.711... Pretty lame if you ask me but, they are in the business of selling bandwidth.

[thisisshanky]

Seth,

Have you confiugred the router as a gateway or a ICT trunk in Callmanager ? Can you uncheck the option, "Wait for far end H245 capabalities" option under the trunk or gateway and see what happens ? Make sure reset the trunk after you update this setting.

Sankar.

PS: please remember to rate posts!

[seth.wilson]

It is configured as a gateway in Call Manager.

I don't see that option to uncheck...

[thisisshanky · Accepted solution]

Duh!

I forgot that you are running CM 3.2 You may not have that option. Not sure if there is a service parameter equivalent to this.

Gateways default to faststart for H323, can you check whether you have an Inbound Fastart option under gateway.

[seth.wilson]

I could not find an equivalent service parameter nor an inbound option on the gateway configuration...

[seth.wilson]

Shanky,

Oh Joy... I found it. It works now. Just for the record I set the Call Manager Service Parameter 'H323FastStartInbound' to True and all calls are completing successfully.

[thisisshanky]

I am glad that worked. Good deal!

[sarenos2006]

Hello, I have a cisco 2651 with ccme 3.3 and I want to route my external calls PSTN through my SIP provider.

parameters form provider are:

sip.provider.com

login and password port 5060

I have configured this in the sip-ua but I have problems.

All of the IP phones I have behond the router try to authenticate.

I only want 1 authentication from the router

Then i need to tell the internal extensions when they have to go to the PSTN via the SIP provider.

Please can you help me? Is there any example?

[thisisshanky]

Do you have the registrar command keyed in. If yes, try removing it under sip-ua mode.

HTH

Sankar

PS: please remember to rate posts!

[seth.wilson]

Hey just an up update for all... Becuase we set up the paramater, it killed some things with incoming calls on other H.323 gateways. Calls would come in fine, but once the call was placed on hold, transferred or conferenced, the PSTN caller would hear dead silence. But, internal users could hear audio from the PSTN user. We went back to setting the parameter to false. The customer is currently planning to upgrade CM to 5.0 in order to move forward. Thanks for the help anyway...

[sarenos2006]

am trying it but i'm very upset because I cannot get it.

I have configured the sip-ua but the only things that are trying to register in the SIP proxy are the IP phones.

I don't want that I want only register one time with the account I have configured in the authentication and then route calls with pattern 9........ to the sip provider

Please I need help

This is my configuration:

dial-peer voice 100 voip

destination-pattern 9........

session protocol sipv2

session target sip-server

codec g711ulaw

!

sip-ua

authentication username xxx password xxx

retry register 10

registrar ipv4:213.162.201.146 expires 60

sip-server ipv4:213.162.201.146

!

!

gatekeeper

shutdown

!

!

telephony-service

max-ephones 2

max-dn 2

ip source-address 192.168.3.1 port 2000

auto assign 1 to 2

user-locale ES

network-locale ES

create cnf-files version-stamp Jan 01 2002 00:00:00

max-conferences 4 gain -6

transfer-system full-consult

!

!

ephone-dn 1 dual-line

number 1 no-reg both

!

!

ephone-dn 2 dual-line

number 2 no-reg both

!

!

ephone 1

mac-address 0013.60C3.CE02

type 7902

button 1:1

!

!

!

ephone 2

mac-address 0012.431E.BF3E

type 7902

button 1:2

!

The logs I receive in the SIP proxy when I try to call to an external number like 964525351 are:

Admission Request BB00000149 : 1@192.168.2.200 => 964525351

$I=BB00000149 exec ifone..sp_gk_start_call '192.168.2.200','','1','964525351','BB00000149','1','Oct 07 2006 19:10:34','',1,'20','','','','DE813E7E-411611D6-80B2C5E4-BE2FDD31','213.162.201.146','2.5.0000@Cisco-SIPGateway/IOS-12.x'

go

BB00000149|STOP|404|>>APPL: AUTH FAILURE/UNKNOWN USER[CODE:2 ACCTID:-1]

Admission Reject BB00000149 : 1@192.168.2.200 => 964525351

I am very frustrated so your help will be great.