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hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get "Invalid CSeq Number" back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave

The file "rtp.c" currently contains the following hack for special-case handling of some Cisco-specific protocol: } else if (payloadtype == 121) { /* CISCO proprietary DTMF bridge */ f = process_type121(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); As I noted in my earlier message, I'm planning to update the code to improve the handling of dynamic RTP pay...

..._test_addr' follows non-static declaration udptl.c:541: warning: 'udptl_debug_test_addr' declared inline after being called udptl.c:541: warning: previous implicit declaration of 'udptl_debug_test_addr' was here udptl.c: In function `ast_udptl_read': udptl.c:643: error: `payloadtype' undeclared (first use in this function) udptl.c:643: error: `timestamp' undeclared (first use in this function) udptl.c:643: error: `hdrlen' undeclared (first use in this function) udptl.c:649: error: `AST_FORMAT_T38' undeclared (first use in this function) udptl.c: In function...

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC