One way audio on CME calls over GRE Tunel to PSTN

[Kunle Ajeigbe]

Hi all,

I ran into some challenges here and need assistance. her is the scenario.

I have 2 Cisco CME installed in 2 different sites. Site A has SIP phones and SCCP phones registered and routes PSTN via an E1 on the CME A  router.

Site B has just been added with SCCP phones registering on the CME B router. PSTN calls are routed thru the E1 on CME A router.

External DIDs are configured on CME A to ring on Phones on Site B when called from PSTN (Incoming calls)

The 2 sites are connected via  VPN over VSAT (GRE Tunnel). internal calls between Site A and B is fine.

Incoming external calls from the PSTN to an extension in Site B routes fine and call is fine.

But when Phone in SIte B places a call to the PSTN, call is connected but returns one way audio. users on the PSTN cannot hear the far end at Site B. but user on SIte B can hear the caller on the PSTN.

I have attached the voip ccapi inout debug & ccsip messages debug.

I can see error SIP/2.0 488 Not Acceptable Media, but don't know how to go around solving this.

Suggestions are welcomed.

[Tristan Cober]

The Invite and 488s you are seeing involving "VaxSIPUserAgent/3.0" appears to be an attack on your router. You should configure tollfraud prevention, permitting only the IP addresses of your routers and phone subnets in your permit list. 

Your One-way audio problem is likely due to routing. I would guess the interface you have your inbound dial-peer at site A bound to is not reachable from the phone subnet at Site B.

[Mohammed al Baqari]

+5 Traistan.

Kunle,

Depending on your CME version, you can configure toll-fraud protection out of the box or use voice source-group feature along with access-lists to define your trusted sources. You are getting SIP INVITES from unknown source while I believe doing some reconnaissance to leak information about your CME.

Your debugs don't include SIP message for calls from site-b to pstn. Please share this. Along with it, we need a traceroute from your site-b cme to site-a pstn router.

[Kunle Ajeigbe]

Mohammed,

I have added some more debugs on the routers during a call.

thanks

[Mohammed al Baqari]

Let's start with the fact that your site-b CME is using H323 for calling. Therefore, your SIP debugs aren't showing any messages for call initiation. To view call setup debugs you need to use 'debug cch323 h225' , 'debug cch323 h245' , 'debug h225 asn'

Now your H323 config is missing source binding which is most likely causing your one-way audio problem.

Apply the following config and test.

interface GigabitEthernet0/2.40
h323-gateway voip interface
h323-gateway voip bind s 192.168.40.1

[Kunle Ajeigbe]

hello Mohammed,

thanks I have added the source binding config but still one way audio persists.

see attached the debugs

[Kunle Ajeigbe]

I have configured toll prevention on both Site A and B router to allow just the voice and router subnets.

I am still getting the one way audio

I have attached the running config on Site B cme and the ccsip messages debug

[Kunle Ajeigbe · Accepted solution]

Hello Thanks Guys,

I have found the problem I omitted the config

"session protocol sipv2"

in the outgoing dial-peer.

thanks very much for the help