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No Inbound audio T1 PRI to VoIP

scott.wysocki
Level 1
Level 1

        I am setting up a new T1 provided by a local ISP.  When call inbound calls arrive from the PRI the call is routed to the proper IP phone but no audio is heard.  Audio works from the VoIP phone to the PSTN side.

My T1 controller counters are clean and isdn status is MULTIPLE_FRAME_ESTABLISHED.

This is a new install so my dial plan is simple:

dial-peer voice 20 pots

destination-pattern 9.T

incoming called-number .T

direct-inward-dial

port 0/0/0:23

But again, my calls are routing inbound and outbound.  I simply can't here audio incoming from the PSTN side.

1 Accepted Solution

Accepted Solutions

Scott,

In my experience 99% of VOIP one way transmission are the
result of IP routing issues.

Do you have mutiple ip addresses on your router.

Have you applied a bind statement for the VOIP protocol.

Regards,
Alex.
Please rate useful posts.

Regards, Alex. Please rate useful posts.

View solution in original post

13 Replies 13

paolo bevilacqua
Hall of Fame
Hall of Fame

Start mentioning system and, version and you're using.

My apologies.

It is an ISR Router 2921 running IOS 15.1 4(M4) with CUCME vers 8.6.

The T1 is terminated on a VWIC3-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk

Di you have complete audio for outbound calls ?           

Altough unlikey, it could be telco problem.

If they are unwilling to come onsite and check, since you have a dual port PRi you could tray conencting with a T1 corss cable and calling from a port to anotehr for testing. It's a bit of work but better than doing nothing while waiting for telco.

I don't have complete (two way) audio for outbound calls. I can only hear audio from CME side, regardless of which side initiates the call.

Is there anything else I can debug before I call the Telco? 

What is the connection VG and  cucm

It it mgcp gw registered to cucm?

Is it just pri and voip dial peers(sip or h323) pointing to cucm?

In the debug isdn q931 did you see any errors?

is it possible to collect sdi traces only with incoming call?

OR IS JUST A CME

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""

Scott,

In my experience 99% of VOIP one way transmission are the
result of IP routing issues.

Do you have mutiple ip addresses on your router.

Have you applied a bind statement for the VOIP protocol.

Regards,
Alex.
Please rate useful posts.

Regards, Alex. Please rate useful posts.

Alex, I am beginning to think that as well.  I just did a capture from a computer running IP Communicator and I see the RTP stream from the computer to the CME router but nothing from CME to the computer. 

I also tested from VoIP to VoIP and it works as expected, the capture shows bi-directional RTP.

My router is setup as a host (single IP) and can ping the subnets my phones are connected to.

Mabe there is an ACL blocking traffic.

I wish it was just an ACL or Routing.  Today I connected directly to the CME router and had the same results.  I still think I am missing something simple, but I see no RTP from the CME to the VoIP endpoint.

Hi Scott,

If possible please post your configuration so we can see the bigger picture and determine whether indeed it is something obvious that a second pair of eyes could identify. It certainly does appear to be an asymmetric problem as you only see RTP in one direction, this would be understandable if you didn't have only IP address on the gateway, so IP connectivity should be straight forward.

Regards
Allan

Sent from Cisco Technical Support iPad App

You may have a faulty DSP. If the router is new, you can ask a replacement as 'DOA' device opening the case to Cisco TAC, with serial number.

Hi,

As Paolo mentioned, DSPs could be attributed to the issue, and errors with the dsps will be logged, show take a moment to review your log output to eliminate or isolate this from possible causes.

Regards
Allan

Sent from Cisco Technical Support iPad App

scott.wysocki
Level 1
Level 1

I appreciate everyones help on this.  It was actually resolved quickly by the TAC.  The fact that ip routing must be enabled for RTP to function may be common knowledge, but I wouldn'thave  thought of this dependency.  Since this router was single-legged (one interface/one IP) there was no reason to turn on ip routing, originally.  Enabling it allowed RTP to function as expected.

I want to mark this is answered so I am picking the first mention of routing.  But thanks to everyone for pitching in.