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

[Sreekanth]

Dear Tony,

These debugs show the signaling between the Asterisk and an X-Lite phone
that is registered with it? I don't see the SIP dialog between the CUCM and
the Asterisk. The problem should lie there.

Regards
Sreekanth

On 5 July 2018 at 17:12, Tony Kasule <timotsmith at gmail.com> wrote:

> Dear Sreekanth.
>
> Thanks for your responses.
>
> Please find my traces attached. The call from cisco to asterisk is fine,
> asterisk to cisco has no audio, i have attached both traces.
>
> regards,
>
>
> On Thu, Jul 5, 2018 at 9:44 AM, Sreekanth <sknth.n at gmail.com> wrote:
>
>> 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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