That's great Tony. Thanks for the info. On 5 July 2018 at 18:57, Tony Kasule <[email protected]> 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 <[email protected]> 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 <[email protected]> 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 <[email protected]> wrote: > >>> > >>> Dear Wilson, > >>> > >>> On 5 July 2018 at 11:56, Tony Kasule <[email protected]> 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 <[email protected]> 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 <[email protected]> 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 > >>>>>> [email protected] > >>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip > >>>>>> > >>>>> > >>>> > >>> > >> > > >
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