UK PSTN International Ringback

DaveX4285_2 · ‎10-18-2011

(SCCP) --> CUCM 8.5 <---(SIP TRUNK)---> 3825 <---(PSTN)PRI

Just brought up a new cluster in the UK and for all international calls I am receiving the same ring back no matter which country is dialed. All ringback is the UK ringback. If Germany is dialed we get UK ringback, Japan = UK ringback. User and Network locale are set to UK and the UK locale is loaded. Under the PRI interface cptone GB is configured. When this site was running CME we recieved the proper ringback depending on the country, but now moving to CUCM that has disappeared. Any help is appreciated.

ADAM CRISP · ‎10-18-2011

hello.

I thing there is a disable early media 180 ringing command under sip-ua , which stops the ringback unless no media is present...

Adam

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DaveX4285_2 · ‎10-18-2011

I noticed this can be disabled on the SIP Profile on CUCM as well. Would this have the same effect?

Thanks

ADAM CRISP · ‎10-18-2011

Aa

I don't thing so. It's the router that terminates the isdn stack, so it is here the sip dialogue starts. I would imagine the setting on the cucm, is for inbound calls.

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DaveX4285_2 · ‎10-18-2011

Thanks. Unfortunately adding this had no effect on the ringback. The relevant part of the configuration is below

!

dial-peer voice 5 pots

trunkgroup 1

description [ Outgoing ]

preference 2

destination-pattern .T

progress_ind alert enable 8

progress_ind connect enable 8

forward-digits all

!

voice-port 0/0/0:15

translation-profile incoming AA-DDI

no vad

cptone GB

timeouts interdigit 4

music-threshold -70

bearer-cap Speech

!

voice service voip

clid substitute name

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

signaling forward unconditional

fax protocol pass-through g711ulaw

modem passthrough nse codec g711ulaw

sip

bind control source-interface GigabitEthernet0/0

bind media source-interface GigabitEthernet0/0

transport switch udp tcp

midcall-signaling passthru

!

ADAM CRISP · ‎10-18-2011

Ok. That's interesting. I wonder what's present on the PRI ?

It might be worth gathering the output of debig ccsip messages and isdn q931 to see whether and when you receive alerting from isdn for UK calls / international and when you send the SIP ringing with sdp message

example:

Oct 18 09:23:56.252: ISDN Se0/1/0:15 Q931: TX -> SETUP pd = 8 callref = 0x367E

Sending Complete

Bearer Capability i = 0x8090A3

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA9839D

Exclusive, Channel 29

Calling Party Number i = 0x2180, '1245'

Plan:ISDN, Type:National

Called Party Number i = 0x91, '44845123456'

Plan:ISDN, Type:International

Oct 18 09:23:56.404: ISDN Se0/1/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0xB67E

...

few moments later

...

Oct 18 09:23:58.356: ISDN Se0/1/0:15 Q931: RX <- ALERTING pd = 8 callref = 0xB67E

Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info

I would expect the gw to send ONLY at this point the 180 with SDP. If this is sent before the progress indicator then it's the Cisco, after then it's possible the telco is sending you UK tones.

In an effort to be helpful, here's the config of our of our UK ISDN to SIP gateways. It's quite a bit simpler than yours.

We get GB ringback for UK and International via early media everywhere else

voice service voip

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw

voice-port 0/0/0:15

cptone GB

bearer-cap Speech

dial-peer voice 10 pots

description POTS talking dial peer for E1 #0

translation-profile outgoing voip-to-pstn

preference 3

destination-pattern .+

incoming called-number .+

direct-inward-dial

port 0/0/0:15

dial-peer voice 997 voip

description incoming voip

rtp payload-type cisco-codec-fax-ind 124

voice-class codec 1

session protocol sipv2

incoming called-number .+

dtmf-relay cisco-rtp rtp-nte

fax-relay ecm disable

fax nsf 000000

fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711alaw

no vad

!

sip-ua

disable-early-media 180

set sip-status 480 pstn-cause 42

set sip-status 500 pstn-cause 42

set sip-status 503 pstn-cause 27

set pstn-cause 3 sip-status 503

set pstn-cause 17 sip-status 500

set pstn-cause 21 sip-status 603

max-forwards 20

sip-server dns:from-pstn.internal.voip.co.uk

Adam

DaveX4285_2 · ‎10-18-2011

Thanks again. Do you have any translations changing the plan and type of the calls by chance. Below is a q931 debug and Im sending ISDN and Unknown where as you are sending ISDN and International.

007131: Oct 18 22:59:41.115 bst: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x0395

Bearer Capability i = 0x8090A3

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98381

Exclusive, Channel 1

Display i = 'Test'

Calling Party Number i = 0x0181, '1007'

Plan:ISDN, Type:Unknown

Called Party Number i = 0x81, '0012489794774'

Plan:ISDN, Type:Unknown

007132: Oct 18 22:59:41.403 bst: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x8395

Channel ID i = 0xA98381

Exclusive, Channel 1

007133: Oct 18 22:59:45.099 bst: ISDN Se0/0/0:15 Q931: RX <- PROGRESS pd = 8 callref = 0x8395

Progress Ind i = 0x8381 - Call not end-to-end ISDN, may have in-band info

Progress Ind i = 0x8482 - Destination address is non-ISDN

007134: Oct 18 22:59:47.883 bst: ISDN Se0/0/0:15 Q931: RX <- ALERTING pd = 8 callref = 0x8395

Progress Ind i = 0x8381 - Call not end-to-end ISDN, may have in-band info

Progress Ind i = 0x8482 - Destination address is non-ISDN

007135: Oct 18 22:59:50.135 bst: ISDN Se0/0/0:15 Q931: RX <- CONNECT pd = 8 callref = 0x8395

007136: Oct 18 22:59:50.139 bst: ISDN Se0/0/0:15 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x0395

UKMCH2VGW001#

ADAM CRISP · ‎10-18-2011

Yes we do.

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DaveX4285_2 · ‎10-18-2011

When you get a chance would you mind sending the translations?

Greatly appreciated.

ADAM CRISP · ‎10-18-2011

Hi.

Not Got live access until tomorrow

But it's a normal voice translation rule followed by something like type any international plan any isdn

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DaveX4285_2 · ‎10-18-2011

No problem. Thats what I created, and it changes the type to international but no difference in ringback.

ADAM CRISP · ‎10-18-2011

ok can you debug ccsip messages and isdn q931 at the same time?

Back tmrrw. Adam

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DaveX4285_2 · ‎10-18-2011

Debug is attached.

ADAM CRISP · ‎10-19-2011

Hi Dave.

You initial invite is offerless meaning that early media isn't going to work.

The 183 Session progress contains all the codecs the GW is configured with configures, but for sure the GW doesn't know where to send the media, or indeed what codec is

You need to get CM to send in invite with SDP. I *think* you can make this happen by making the SIP trunk use an MTP.

If the GW is part of your network I'd be inclined to try the following

1. On CM tick the box to use an MTP that is available in the SIP trunk's region.

re-test

2. If this fails or if you want to perfect your media flow, configure your GW router to be an MTP resource for CM and then make this MTP available for this SIP trunk only.

Interesting one!

Adam

DaveX4285_2 · ‎10-19-2011

Thanks Adam, I will look at making these changes after hours.

Regarding your 2nd remark. This is currently a single site Pilot deployment, would there be any benefits in configuring the MTP local on the gateway if its all at the same site. Obviously this would be preferred but at this point the gateway has limited DSP resources and I had been utilizing the CUCM's resources. I will make the change after hours and let you know the result.