First of all, switch to SIP, then make a test call and collect logs in CUCM, 2800 and Asterisk. Just compare the codec offered in the SDP. Modify the CUCM/2800/Asterisk so they all have the same codec, I.E g711alaw AND ulaw, 20ms sample rate etc. remove any other codecs not used.
You want to avoid using MTP in CUCM as resources are limited. You can use software MTP in 2800 but it just adds complexity to solve a simple problem. On Mon, May 11, 2015 at 5:16 PM, Brian Meade <bmead...@vt.edu> wrote: > If you want us to be able to figure out your H.323 negotiation problem > without codec transparent in place, you need to provide these debugs: > debug h225 asn1 > debug h245 asn1 > > On Mon, May 11, 2015 at 6:33 AM, s m <sam.gh1...@gmail.com> wrote: >> >> thank you Ryan, >> >> i have no problem with sip and it is ok (although i do not know it has MTP >> or not). i think transcoder may not needed because as i know, it translate >> two different codecs to each other but in my scenario, both side uses >> g711alaw. please let me know if i misunderstand it. >> >> >> On Mon, May 11, 2015 at 11:31 AM, Ryan Huff <ryanh...@outlook.com> wrote: >>> >>> The reason that is happening is due to media negotiation failure as you >>> mention (both call legs are not offering the same codec capabilities). In >>> that exact configuration, you would need a transcoder (which you could run >>> on the router if you have enough DSP). >>> >>> Are you sold on h323 or can you do a full SIP trunk (with MTP) between >>> cucm and asterisk? >>> >>> Thanks, >>> >>> Ryan >>> >>> >>> >>> -------- Original Message -------- >>> From: s m <sam.gh1...@gmail.com> >>> Sent: Monday, May 11, 2015 12:25 AM >>> To: Ryan Huff <ryanh...@outlook.com> >>> Subject: Re: [cisco-voip] how codec transparent works? >>> CC: cisco-voip@puck.nether.net >>> >>> hello Ryan, >>> >>> thank you for your reply. without codec transparent, my phone rings but >>> when i answer i have no voice and it hangs up after 5 seconds. asterisk says >>> "no answer". this is so strange for me. i think media negotiation failed, >>> right? is there any hint to have h323 trunk to asterisk with specific codec >>> (not transparent one)??? >>> >>> On Sun, May 10, 2015 at 4:32 PM, Ryan Huff <ryanh...@outlook.com> wrote: >>>> >>>> Codec transparent just passes sdp through to the other call leg without >>>> trying to do media negotiations. >>>> >>>> So without codec transparent, what happens? >>>> Thanks, >>>> >>>> Ryan >>>> >>>> >>>> >>>> -------- Original Message -------- >>>> From: s m <sam.gh1...@gmail.com> >>>> Sent: Sunday, May 10, 2015 01:19 AM >>>> To: cisco-voip@puck.nether.net >>>> Subject: [cisco-voip] how codec transparent works? >>>> >>>> hello everybody, >>>> >>>> anybody knows how codec transparent works? >>>> >>>> i have a strange problem. i want to set h323 trunk between asterisk and >>>> cisco 2800. it only works when i set codec transparent in dial-peer nodes. >>>> show commands in cisco shows that i have a call with g711alaw but if i set >>>> codec g711alaw in dial-peers, i do not have any success call. i know it is >>>> codec compatibility problem. is there any difference between g711 codecs >>>> which cisco and asterisk utilize? what happened when codec is set to >>>> transparent? dose anyone know anything about it? >>>> >>>> thanks is advance >>>> SAM >>> >>> >> >> >> _______________________________________________ >> cisco-voip mailing list >> cisco-voip@puck.nether.net >> https://puck.nether.net/mailman/listinfo/cisco-voip >> > > > _______________________________________________ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > _______________________________________________ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
