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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