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

[Sreekanth]

That's great Tony. Thanks for the info.

On 5 July 2018 at 18:57, Tony Kasule <timotsmith at gmail.com> wrote:

> 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
> >>>>>>
> >>>>>
> >>>>
> >>>
> >>
> >
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20180705/c97a82dc/attachment.html>