[cisco-voip] One Way Voice on Cisco-Asterisk

Tony Kasule timotsmith at gmail.com
Thu Jul 5 01:22:24 EDT 2018


Dear Friends,

I have CUCM 11 and Cisco gateway at my organisation where I am trying to
add a small asterisk call center.

I created a SIP trunk between CUCM and Asterisk 15.4  on Centos 7 and also
did the same at the gateway. When I call from the PSTN to a dial-peer that
is mapped to asterisk, the call goes through well and we each each other.
However, when I call from asterisk to the PSTN, The call goes through but
there is total silence.

Same issue with asterisk-CUCM. When I call from the Call manager to
asterisk, its fine but asterisk to cisco extension, there is no audio on
answering the call.

I have been perplexed by this scenario. I extensively read online, turned
MTP on and off (at the sip trunk in CUCM), tried nat=yes and nat=no at
asterisk side etc but no joy yet.

Has anyone else experiences this and any pointers on how to have it
resolved?

Thank you so much.

Timothy
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20180705/fdee07a4/attachment.html>


More information about the cisco-voip mailing list