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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