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

[Sreekanth]

Tony,
Are you seeing a complete SIP dialog between Asterisk and CUCM when making
the calls from Asterisk towards the phone? Are the IPs and ports advertised
in the SDP correct?

I would start by taking a packet capture at the gateway or asterisk to see
if 2 way RTP is flowing between them. If you enable MTP then you can also
enable a pcap on the CUCM where the MTP is located.
This would help isolate where the packets are being lost.

Regards
Sreekanth

On 5 July 2018 at 10:52, Tony Kasule <timotsmith at gmail.com> wrote:

> 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
>
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