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

[Tony Kasule]

Dear Sreekanth,

Thank you so much for helping me.

My mac laptop where is soft phone is installed has an inbuilt firewall
that was enabled. While troubleshooting, I couldn't ping my laptop
from the call manager so I decided to disable the firewall and Voila!
Everything worked. So, the issue was the firewall on my laptop.
Unfortunately, I hadnt tested with a real phone or a phone on another
laptop so I couldnt figure that out yesterday.

Thanks!

Regards,
Tony

On Thu, Jul 5, 2018 at 4:02 PM, Sreekanth <sknth.n at gmail.com> wrote:
> 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
>>>>>>
>>>>>
>>>>
>>>
>>
>