Inbound FXO Calls Ring Once, then Dial Tone; MGCP

Unanswered Question
Dec 3rd, 2008

I have a 2811 router configured as a MGCP gateway to a CUCm 6.0 server. The FXO ports are configured and registered according to CUCM. Inbound call ring the correct port via a "show voice port summary" but it only rings once and then you get dial tone.

The port is configured to ring extension 8208 which is a CTI route point that forward all to voicemail on a Unity server. There is a call handler on Unity with that number.

Calls to other sites in this environment can transfer to the 8208 auto attendant without a problem, so I'm assuming that the CTI route point and call handler are configured correctly.

Any help would be appreciated.

I have this problem too.
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Jaime Valencia Wed, 12/03/2008 - 11:37

u sure the MGCP config on the router is right??

do you have a dial peer with application mgcpapp for that FXO?? FXO registered??



if this helps, please rate

gmgarrian Wed, 12/03/2008 - 11:44

Here's the port and MGCP portion of the config. The port is registered to the correct server IP and is showing the correct IP from the gateway:


voice-port 0/1/0

timing hookflash-out 50

description LINE ONE


dial-peer voice 999010 pots

service mgcpapp

port 0/1/0


ccm-manager fallback-mgcp

ccm-manager redundant-host

ccm-manager mgcp

ccm-manager music-on-hold

ccm-manager config server

ccm-manager config



mgcp call-agent 2427 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode out-of-band

mgcp rtp unreachable timeout 1000 action notify

mgcp modem passthrough voip mode nse

mgcp package-capability mf-package

mgcp package-capability rtp-package

no mgcp package-capability res-package

mgcp package-capability sst-package

no mgcp package-capability fxr-package

mgcp package-capability pre-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp rtp payload-type g726r16 static

mgcp bind control source-interface FastEthernet0/0.20

mgcp bind media source-interface FastEthernet0/0.20


mgcp profile default


Jesse Drohen Wed, 12/03/2008 - 14:56

Is this gateway configured for SRST or setup with H323 dial peers? If so and you shut down those dial peers does it work? "debug voip dialpeer inout" is a good one to tell what dial peers you are hitting.


gmgarrian Wed, 12/03/2008 - 15:09

I'm currently working with TAC on this. It appears that after the initial single ring, you get a dial tone. If you let it go, after 15 seconds it will connect as desired. Now they're trying to figure out why that is happening and how to fix it.

Any ideas on that?

seaman0555 Thu, 12/04/2008 - 10:30

Do you have these statements in your router?

voice-card 0

no dspfarm

dsp services dspfarm



voice call send-alert

voice call carrier capacity active

voice rtp send-recv


voice service voip

allow-connections h323 to sip


Check your working sites and see what you have there.

gmgarrian Thu, 12/04/2008 - 11:02

The config I posted is exactly the same for the working sites. I did get this working by using a pilot number in the 4XXX range instead of the 8XXX range. The other sites are configured for 8XXX range AA pilots but for some reason this site would wait for the inter-digit timeout if I used it here.

It's working correctly now with the 4XXX pilot but I'd still like to figure out why the 8XXX won't work. I had TAC close the case since we're up and running.

Any ideas what would block the 8XXX range?

acjarvis Mon, 06/01/2009 - 18:31

I've been looking through posting for awhile and your problem seems to be the exact problem I am having.

Since your posting in December have you ran across this again? If so, do you have a fix?

I've been running through my CCM and gateways configs but nothing is popping out.




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