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One-way audio on E1

Ruslan Amrahov
Level 1
Level 1

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?

4 Replies 4

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 yes

https://supportforums.cisco.com/blog/12122171/cisco-support-community-helps-kiva

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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

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

Please rate all helpful posts "The more you help the more you learn"

Murali Dharan
Level 1
Level 1

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.