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...show channels Chan Extension Context Language MusicOnHold pseudo incoming it 1 incoming 2 incoming *** *CLI> show dialplan [ Context 'incoming' created by 'pbx_config' ] '+390372xxxxxx' => 1. Dial(SIP/cisco2) [pbx_config] '00372xxxxxx' => 1. Dial(SIP/cisco2) [pbx_config] 'xxxxxx' => 1. Dial(SIP/cisco2) [pbx_config] '0372xxxxxx' => 1. Dial(SIP/cisco2)...

...between the two ciscos... therefore, once the communication is accomplish it hangs up. I had to make available both g729a anf g729 in my oh323.conf, since the first cisco appears to use that codec. I tried all the combinations successfully, since cisco->asterisk->sip works ok and asterisk->cisco2 works too. I am sure this works since asterisk is transcoding can not make the rtp to flow directly between the two clients. Does somebody knows how to deal with this? Thanks, Alito

...allow=ulaw firstdigittimeout = 16 digittimeout = 8 diggittimeoutchar = # rtptos = 184 echocancel = on tos = 0x68 mwilamp = on mwioncall = on autoanswer_tone = 0x32 remotehangup_tone = 0x32 dtmfmode = inband dnd = on [devices] type = 7960 description = Cisco2 tzoffset = 0 trustphomeip = no autologin = 105 device => SEP00036B0123 [lines] id = 104 pin = 1234 label = 104 description = Cisco1 context = internal ;callwaiting = 1 incominglimit = 2 mailbox = 104 vmnum = 500 cid_name = Cisco1 ci...

...; ;[cisco] type=friend username=9873 secret=pwd ;nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away defaultip=192.168.0.5 mailbox=9873 ;[cisco2] type=friend username=9874 secret=pwd nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away defaultip=192.168.0.10 mailbox=9874 extensions...

...; RTP media stream (audio) to go directly from ; the caller to the callee. Some devices do not ; support this (especially if one of them is ; behind a NAT). ;defaultip=192.168.0.4 ;[cisco2] ;type=friend ;username=cisco2 ;fromuser=markster ; Specify user to put in "from" instead of callerid ;fromdomain=yourdomain.com ; Specify domain to put in "from" instead of callerid ; fromuser and fromdomain are used when Asteri...

...a stream (audio) to go directly from > ; the caller to the callee. Some devices do not > ; support this (especially if one of them is > ; behind a NAT). > ;defaultip=192.168.0.4 > > ;[cisco2] > ;type=friend > ;username=cisco2 > ;fromuser=markster ; Specify user to put in "from" instead of callerid > ;fromdomain=yourdomain.com ; Specify domain to put in "from" instead of callerid > ; fromuser and fromdo...