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

[Sreekanth]

Dear Wilson,

On 5 July 2018 at 11:56, Tony Kasule <timotsmith at gmail.com> wrote:

> 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.
>

MTPs are only required for functions such as dtmf mismatch and
packetization mismatches between the 2 legs, or if you'd like to force
Early Offer. The CUCM will invoke an MTP on its own if the call requires it.


> 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?
>

What is the remote IP address and port? Yes this means that packets are not
making it from remote end to the phone.


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

Which message had the Content length 0? Can you paste a snippet here?


>
> 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.
>
>
If you could paste the entire SIP dialog debug here, we can take a look to
see what exactly is going on in the exchange.


> Thanks for your help in advance.
>
> Regards,
> wilson
>
>
>
Thanks
Sreekanth


> 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
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>
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