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New Member

SIP TRUNK 503 Service Unavailable

Hi

We have an issue with calls from PSTN. When we received a call from PSTN the caller hears an IVR and then press the option "2" then the caller should be transfered to extension 96537, that is another IVR, but the caller hears ringback and then the call is dropped.

Our call flow is like this:

PSTN > R2 > GW > SIP > CVP > SIP > CUCM > H323 > SME > 96537

Atteched the call log from GW, in that log we saw this message:

NOTIFY sip:10.1.1.13:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 150.212.102.250:5060;branch=z9hG4bK3E22427

From: <sip:5556236510@150.212.102.250>;tag=11A617A4-798

To: <sip:3007@10.1.1.13>;tag=ds7d41310a

Call-ID: F07DEE62-669B11E3-8E71C240-5FB154C1@150.212.102.250

CSeq: 103 NOTIFY

Max-Forwards: 70

Date: Tue, 17 Dec 2013 21:49:42 GMT

User-Agent: Cisco-SIPGateway/IOS-15.3.3.M

Event: refer

Subscription-State: terminated;reason=noresource

Contact: <sip:5556236510@150.212.102.250:5060>

Content-Type: message/sipfrag

Content-Length: 35

SIP/2.0 503 Service Unavailable

039256: *Dec 17 21:49:42.639: //0/000000000000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 150.212.102.250:5060;branch=z9hG4bK3E11BE0

To: <sip:3007@10.1.1.13>;tag=ds7d41310a

From: <sip:5556236510@150.212.102.250>;tag=11A617A4-798

Call-ID: F07DEE62-669B11E3-8E71C240-5FB154C1@150.212.102.250

CSeq: 102 NOTIFY

Content-Length: 0

Allow-Events: refer

Allow-Events: kpml

Allow-Events: cvp-transfer

I hope you can help me.

Regards.

5 REPLIES
New Member

SIP TRUNK 503 Service Unavailable

A correction from my last messege

The call flow is like this:

PSTN > R2 > GW > SIP > CVP > SIP > GW >  CUCM > H323 > SME > 96537

Sorry for my error.

Regards.

SIP TRUNK 503 Service Unavailable

Hi CCoria,

Can you please share your SIP trunk snapshot from CUCM?

Also share the configuration of all the gateways.

Also please clarify your call flow once as CUCM connectivity is not cleared yet.(Is R2 & GW is one device?).

Regards,

Nishant Savalia

Regards, Nishant Savalia

SIP TRUNK 503 Service Unavailable

Hi,

it seems codec issue. have you configured 'early-offer forced' command in GW pointing to CUCM.

If I understand correctly, the inbound call coming from PSTN is transferred by CVP to GW and then to CUCM for extn: 96537.

The SIP Invite message from CVP to GW has no SDP. Also the SIP Invite from GW to CUCM has no SDP as it might be missing the EO Forced command.

in case if it is not configured previously, you can enable the Forced EO in 2 ways.

1) under the "voice service voip" -> "sip" configuration, configure 'early-offer forced'

2) under the dial-peer, the command will be "voice-class sip early-offer forced" . This will only take effect when  the call hits this dial-peer while the previous command under "voice service voip" will have global effect. SDP info would be from the codecs configured under outgoing dial-peer.

Please ensure you configure the Outgoing Dial-peer=96537 with voice class codec consisting g711a, g711u & g729 codecs

when the GW sends the INVITE without SDP to CUCM, the CUCM sends the 200 OK message with G729 codec in the sdp.

039123: *Dec 17 21:49:40.615: //2100/F07CB6128E6C/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 150.212.102.250:5060;branch=z9hG4bK3DE470

From: <5556236510>;tag=11A64094-1CAF

To: <96537>;tag=8933~4edf650c-7a65-470e-8db1-1714957ea965-46001607

Date: Tue, 17 Dec 2013 21:56:50 GMT

Call-ID: F6BD0C3F-669B11E3-8E8AC240-5FB154C1@150.212.102.250

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

Allow-Events: presence

Supported: replaces

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires:  1800;refresher=uas

Require:  timer

P-Asserted-Identity: "Opcion 3 SUN" <96537>

Remote-Party-ID: "Opcion 3 SUN" <96537>;party=called;screen=yes;privacy=off

Contact: <96537>

Content-Type: application/sdp

Content-Length: 257

v=0

o=CiscoSystemsCCM-SIP 8933 1 IN IP4 150.211.101.253

s=SIP Call

c=IN IP4 10.2.1.5

b=TIAS:64000

b=AS:64

t=0 0

m=audio 24652 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=ptime:20

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

>> is the codec negotiation between CUCM & GW configured as G729? Please crosscheck the SIP trunk configuration & region settings for this.

>> The call is disconnected with "Reason: Q.850;cause=65" indicating "no matching codec".

>> As it is CVP, we may need to configure the dial-peers with G711ulaw towards CVP.

//Suresh Please rate all the useful posts.
New Member

SIP TRUNK 503 Service Unavailable

Hi Sureshsub2

Our codecs should be like this:

All devices behind CUCM should talk G711ulaw

CUCM and SME should talk G729

This are our dial-peers to CVP and ext. 96537

dial-peer voice 3009 voip

description CVP SIP Comprehensive dial-peer

destination-pattern 3009

session protocol sipv2

session target ipv4:10.1.1.13

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 13009 voip

description CVP SIP Comprehensive dial-peer

preference 1

destination-pattern 3009

session protocol sipv2

session target ipv4:10.2.1.9

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 96537 voip

destination-pattern 96537

session protocol sipv2

session target ipv4:150.211.101.253

dtmf-relay rtp-nte h245-signal h245-alphanumeric

voice-class codec 1

no vad

And my voice-class codec is this:

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729br8

codec preference 3 g729r8

150.211.101.253 is CUCM 9.1

10.1.1.13 is CVP CALL/VXML SERVER A

10.2.1.9 is CVP CALL/VXML SERVER B

Regards.

César Coria

SIP TRUNK 503 Service Unavailable

Did you try with the early-offer forced command in GW? Can you pls share the compete GW config? Also do you have transcoders in CUCM for CVP to SME calls?

//Suresh Please rate all the useful posts.
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