ā05-16-2014 03:30 AM - edited ā03-16-2019 10:49 PM
Dear Community
Here is a scenario:
<Analog_Phone>----<H323 GW>----<Gatekeeper>---<CUCM>-----<MGCP>----E1---<PSTN>
When the call is made from any of the IP Phones, registered on the CUCM, the call succeed. But, when I do a call from Analog phone I experience one-way audio so the calling party does not hear anything. I've checked the routing, H323 gateway bind-scr interface is pingable by the MGCP gateway and CUCM. What could be the issue?
ā05-16-2014 03:37 AM
Hi.
Can you please share a show voip rtp connection during an active call?
Thanks
Regards
Carlo
Hi.
In which way they are unable to configure it?
Adding a new destination to an existing one?
..or they cannot see the option to add a new destination.
Once they click to "add new" , add the new destination and selecting the right profile, they should be able to activate it by putting a flag on "reach me anywhere" option and then "save"
Please let me know
Regards
Carlo
Please support CSC helps Kiva
https://supportforums.cisco.com/blog/12122171/cisco-support-community-helps-kiva
ā05-16-2014 03:49 AM
Hi Carlo!
show voip rtp connections
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 47176 47175 28684 17000 172.16.94.1 10.200.200.20
ā05-16-2014 04:57 AM
Hi.
In this case try to ping 10.200.200.x subnet from 172.16.94.x subnet and see if they can reach each other.
HTH
Regards
Carlo
ā05-16-2014 04:15 AM
In my experience, network miss routing causing most of the one way audio issue. Are you able to ping the CUCM and MGCP from H323 gateway?. IP 10.200.200.20 is in remote IP of show rtp connection output which device IP it is?. Check the voice codec config in dial-peer and voice-port config of H323 gateway.
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