12-03-2008 11:23 AM - edited 03-15-2019 02:53 PM
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.
12-03-2008 11:37 AM
u sure the MGCP config on the router is right??
do you have a dial peer with application mgcpapp for that FXO?? FXO registered??
HTH
java
if this helps, please rate
12-03-2008 11:44 AM
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 10.2.0.2
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 10.2.0.1
ccm-manager config
!
mgcp
mgcp call-agent 10.2.0.1 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
!
12-03-2008 02:56 PM
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.
Jesse
12-03-2008 03:09 PM
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?
12-04-2008 10:30 AM
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.
12-04-2008 11:02 AM
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?
06-01-2009 06:31 PM
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.
Thanks,
Aaron
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