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 >>>>>> >>>>> >>>> >>> >> > _______________________________________________ cisco-voip mailing list [email protected] https://puck.nether.net/mailman/listinfo/cisco-voip
