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Beginner

SIP Phone via AT&T connect to CUCM occur one way audio

The customers have multiple sites,one site has CallManager 10.5 ,the ip phones of other sites register on this CUCM,

every site connect each other via AT&T Private network;

now has a problem,the sip phones eg:7821&8831 of other sites that has no CallManager  call out vi local pstn occur one way audio,

7821 can hear ovoice audio from pstn ,but outside can't hear voice audio from 7821;

7945 register on the CUCM use SCCP  is ok;

can you help me ?

5 REPLIES 5
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Rising star

Can you connect a desktop/laptop to PC port of phone and  take wire shark traces, then you will get to know in case of any port blockage.

Suresh 

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Hi Suresh Kumar

I have captured via wireshark , the audio from ip phone 7821 transfer to voice gateway;

voice gateway can receive the audio stream , but can not transfer out;

I have did two tests:

1、let the phone register on local voice gateway , all call is OK;

2、configure another callmanager in local,the new callmanager as one of the cluster ,then let all phones register on the local callmanager, all call is OK;

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Which gateway this ? I mean mgcp or h323 or sip ? Seems issue with codec mismatch. 

Suresh

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h323

CallManager Site <------------AT&T------------>H.323 Gateway Site<-------------->E1

                                                                                    I

                                                                                    I

                                                                                    I

                                                                         SIP Phone 7821

if codec mismatch , can the call establish successfully ?

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It can be, have u verified h323 binding signaling in gateway.

Please try to put this command "voice rtp send-rcv" in gateway and make a call..and pl attach 'debug cch323 rtp' logs And can u attach sh run of gateway as well.

Suresh