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asharsidd
Beginner

SIP Calls ringback issue from PSTN

I am looking into this for customer where users dialling out via a SIP gateway are getting intermittent ringback issue. Sometimes they get ringback, sometimes the call connects direct without any ringing tone. This is getting a bit annoying for customer.

I have logged it with provider and they are sending 183 messages instead of 180. I know 183 messages are cut-through and ringback depends on far-side gateway and therefore I requested them to send us 180 ringing so I can disable it at the gateway like as follows:

sip-ua

disable-early-media 180

!

But they are saying it is us who are requesting 183 messages...is this true? how can I change this behaviour? also they are saying they are sending back ringback but from their traces it appears the call is not even waiting for ringback and is getting connected with 200 OK. Is there a timer issue?

User >>>> CCM >>>> SIP-GW ----> Provider >>> PSTN >>>> Phone

<<<<<<<NO ringback

Here are few configs:

voice-card 0
dsp services dspfarm
!
!
voice call send-alert
voice call convert-discpi-to-prog
!
voice service voip
allow-connections sip to sip
supplementary-service h450.12
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
sip
  rel1xx disable
  header-passing
  midcall-signaling passthru

!

interface GigabitEthernet0/0
description *** METH (SIP Trunk) to provider ***
ip address 83.166.x.x 255.255.255.0
ip access-group SIP-Security-In in
ip access-group SIP-Security-Out out
no ip redirects
no ip unreachables
no ip proxy-arp
ip flow ingress
duplex full
speed 100
no cdp enable
no mop enabled

!

!

voice class codec 1
codec preference 3 g729r8
codec preference 8 g711alaw
!
!

!
dial-peer voice 10 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming SIP-CALLS-IN
preference 1
redirect ip2ip
session protocol sipv2
session target ipv4:83.166.x.x
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte

no vad
!

dial-peer voice 20 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing SIP-CALLS-OUT
max-conn 200
destination-pattern 9.T
progress_ind setup enable 3
progress_ind progress enable 8
progress_ind connect enable 8
redirect ip2ip
session protocol sipv2
session target ipv4:83.166.x.x
voice-class codec 1
no voice-class sip asserted-id
voice-class sip dtmf-relay force rtp-nte
no voice-class sip early-offer forced
dtmf-relay rtp-nte
clid strip pi-restrict all
no vad
!

sip-ua
disable-early-media 180
retry invite 2
retry bye 2
retry cancel 2
!

IOS:

Cisco IOS Software, C2951 Software (C2951-UNIVERSALK9-M), Version 15.0(1)M1, RELEASE SOFTWARE (fc1)

c2951-universalk9-mz.SPA.150-1.M1.bin

echnology Package License Information for Module:'c2951'

----------------------------------------------------------------
Technology    Technology-package          Technology-package
              Current       Type          Next reboot 
-----------------------------------------------------------------
ipbase        ipbasek9      Permanent     ipbasek9
security      None          None          None
uc            uck9          Permanent     uck9
data          None          None          None

Could this be related to Bug Id: CSCsw34198 ?

Debugs have been attached. Any help will be appreciated.

5 REPLIES 5
Marcel Ammann
Participant

Hello,

i don't know if this could resolve the issue but what happend if you set the callmanager service parameter

CUCM Administrator>System>Service Parameters

Change the

SIP Rel1XX Enabled from the default of False to True

Kind regards,

Marcel Ammann

Hi,

I didn't check the service parameter but under Voice service voip SIP I did enable rel1xx that did not make any difference.

asharsidd
Beginner

Can someone please suggest how we can resolve this issue.

Thanks

asharsidd
Beginner

Amy suggestions?

Turn off the Early Offer in CUCM  -  Uncheck the MTP

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