11-05-2007 06:54 AM - edited 03-15-2019 07:03 AM
Whenever I dial into a phone or have the phone dial out it's always a one way conversation. I can never talk on the IP phone and have the other end hear it. The other end can always talk and it will come through the IP phone. The network topology is a CCM box then routes to a ASA then to a MGCP gateway. Any ideas or tips on what to look for? It's almost as if the firewall is blocking a port or something but I have everything opened between the two boxes.
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11-07-2007 03:16 AM
The following is the VoIP call leg extract from the MGCP_Log2:-
MDCX 1580 S0/SU2/DS10/19@generaldisarray.part.n_codec_list(647):[lvl=0]begin
Nov 7 00:12:09.223: //-1/xxxxxxxet MGCP 0.1
C: D000000001a3a96b000000F580000066
I: 3
X: 13
L: p:20, a:G.729, s:off, t:b8
M: sendrecv
R: D/[0-9ABCD*#]
S:
Q: process,loop
v=0
o=- 3 0 IN EPN S0/SU2/DS1-0/19@generaldisarray.part.net
s=Cisco SDP 0
t=0 0
m=audio 24896 RTP/AVP 18
c=IN IP4 192.168.30.250
<---
You can see that the compression codec algorithm option is a:G.729 is only G.729, I would have expected something like a:PCMU;PCMA;G726-32.
When the call is established can run 'show voice call summary' and 'show voice dsp' and post the output.
Also can you confirm that the call agent name tweak is in fact the hostname of your CCM on IP address 192.168.30.250?
Can you amend the mgcp call-agent command to 'mgcp call-agent 192.168.30.250 2427 service-type mgcp version 0.1'
Also when you configured the MGCP bind commands did you initiate a reset of the gateway through CCM?
Please could you also post the output of the 'show ccm-manager' command.
Regards
Allan.
11-07-2007 03:28 PM
I did what you requested and the codec is still G.729. Also, tweak.part.net is 192.168.30.250. A did some tcpdumps and I noticed I wasn't seeing RTP data from the phone. Looking at the network configuration settings I noticed that a default gateway wasn't being picked up. After restarting the DHCP services in the CUCM box I reset the phones and they picked up the default gateway. It now works, thank you a ton for the help you provided. It was the show voice commands that lead to me to find out about the RTP packets which lead me to do a dump.
Rob
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