You should also be able to set it in the gateway configuration, I believe:
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Variables
If that works then that'll probably make it far easier to manage as you won't need to put in any logic to alter the dialstring.
-Steve
OK then!Looking at my call logs, it seems to mostly happen with one particular carrier. Can I set verbose_sdp=true in the bridge string? e.g. "[verbose_sdp=true]sofia/gateway/... "?Thanks,-AviOn Tue, Sep 13, 2011 at 6:28 PM, Michael Collins <msc@freeswitch.org> wrote:For the record, line #107 contains the codecs (and other stuff) that FS offered:m=audio 16480 RTP/AVP 8 0 9 101 13
The "8 0 9 101 13" represent the codecs (and other stuff). What do those numbers mean? This wiki page has the list:The SDP spec says that for non-dynamic payload numbers you are NOT required to have the "a=rtpmap" lines. This makes sense because those payload numbers don't change meaning until you get up to 96 and higher. The issue is that some devices are broken and require the "a=rtpmap" lines for the payload types below number 96. The verbose_sdp chan var mentioned by KK tells FS to add all those a=rtpmap lines for the non-dynamic payload types. You should only set this var when absolutely necessary because those verbose SDP's can cause the packets to be larger than the allowable MTU size.Dontcha just *love* SIP/SDP interop? :)-MCOn Tue, Sep 13, 2011 at 8:01 AM, Avi Marcus <avi@avimarcus.net> wrote:
I see a failed bridge to my second carrier. The sip_local_sdp_str doesn't show PCMU in it - did FS not offer any codecs?PCAP: http://ge.tt/95tFji7Can someone tell me what's going on here?-Avi
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