<div dir="ltr">Dear Wilson,<br><div class="gmail_extra"><br><div class="gmail_quote">On 5 July 2018 at 11:56, Tony Kasule <span dir="ltr"><<a href="mailto:timotsmith@gmail.com" target="_blank">timotsmith@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Dear <span style="font-size:12.8px;text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">Sreekanth,</span><div><span style="font-size:12.8px;text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><br></span></div><div><span style="font-size:12.8px;text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">Thanks for your response.</span></div><div><span style="font-size:12.8px;text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><br></span></div><div><span style="font-size:12.8px">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.</span></div></div></blockquote><div><br></div><div>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.<br></div><div> <br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><span style="font-size:12.8px"><br></span></div><div><span style="font-size:12.8px">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?</span></div></div></blockquote><div><br></div><div>What is the remote IP address and port? Yes this means that packets are not making it from remote end to the phone. <br></div><div> <br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><span style="font-size:12.8px"><br></span></div><div><span style="font-size:12.8px">In the asterisk SIP debug, i saw Content-Length: 0 in the exchanges but I wonder what would cause that.</span></div></div></blockquote><div><br></div><div>Which message had the Content length 0? Can you paste a snippet here?<br></div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><span style="font-size:12.8px"><br></span></div><div><span style="font-size:12.8px">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.</span></div><div><span style="font-size:12.8px"><br></span></div></div></blockquote><div><br></div><div>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.<br></div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><span style="font-size:12.8px"></span></div><div><span style="font-size:12.8px">Thanks for your help in advance.</span></div><div><span style="font-size:12.8px"><br></span></div><div><span style="font-size:12.8px">Regards,</span></div><div><span style="font-size:12.8px">wilson</span></div><div><div class="h5"><div><span style="font-size:12.8px"><br></span></div><div class="gmail_extra"><br></div></div></div></div></blockquote><div><br></div><div>Thanks</div><div>Sreekanth<br></div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div class="h5"><div class="gmail_extra"><div class="gmail_quote">On Thu, Jul 5, 2018 at 8:30 AM, Sreekanth <span dir="ltr"><<a href="mailto:sknth.n@gmail.com" target="_blank">sknth.n@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Tony,</div><div>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?</div><div><br></div><div>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. <br></div><div>This would help isolate where the packets are being lost. <br></div><div><br></div><div>Regards</div><div>Sreekanth<br></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="m_-2423941198171845018h5">On 5 July 2018 at 10:52, Tony Kasule <span dir="ltr"><<a href="mailto:timotsmith@gmail.com" target="_blank">timotsmith@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="m_-2423941198171845018h5"><div dir="ltr"><div class="gmail_extra">Dear Friends,</div><div class="gmail_extra"><br></div><div class="gmail_extra">I have CUCM 11 and Cisco gateway at my organisation where I am trying to add a small asterisk call center. </div><div class="gmail_extra"><br></div><div class="gmail_extra">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.</div><div class="gmail_extra"><br></div><div class="gmail_extra">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.</div><div class="gmail_extra"><br></div><div class="gmail_extra">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.</div><div class="gmail_extra"><br></div><div class="gmail_extra">Has anyone else experiences this and any pointers on how to have it resolved?</div><div class="gmail_extra"><br></div><div class="gmail_extra">Thank you so much.</div><div class="gmail_extra"><br></div><div class="gmail_extra">Timothy</div></div>
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