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

[Tony Kasule]

Dear Sreekanth,

Thanks for your response.

When I enabled MTP on the cisco call manager, I could no longer get audio
even o the cisco to asterisk calls (that were working before). Audio was
restored when I disabled MTP option on the call manager. I later came to
learn that the MTP option is not required when using he same codec both
sides.

I also checked on the cisco 7945 phone and during the call from asterisk to
cisco (which has no audio) and I noticed that Sender Packets is counting
and incrementing but Receiver Packets is 0. Does this mean that the cisco
phone is not receiving any packets, and if so, why?

In the asterisk SIP debug, i saw Content-Length: 0 in the exchanges but I
wonder what would cause that.

Lastly, the SDP IP addresses appear to be ok. I only see the 2 IPs of the
devices communicating. I also went to asterisk's rtp.conf and disabled
strictrtp but no joy yet. Asterisk to cisco calls are silent yet cisco to
asterisk calls are ok.

Thanks for your help in advance.

Regards,
wilson


On Thu, Jul 5, 2018 at 8:30 AM, Sreekanth <sknth.n at gmail.com> wrote:

> 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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>>
>>
>
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