On Tue, Oct 16, 2012 at 1:31 PM, Richard Kenner <[email protected]> wrote:
> We recently set up a SIP trunk between an office in NY running Asterisk and > an office in Paris (running Alcatel). All works fine if a SIP phone on the > NY system talks to the Paris PBX. But if something on DAHDI (a PRI or > MeetMe) talks to the Paris PBX, there's a low-volume crackling. This isn't > clipping because it also occurs when there's no legitimate sound. It's > sort of a mild version of what you used to get when a POTS pair had a > ground short. This occurs no matter what size originates the call. > > pings show round trip times of around 100ms, ranging from around 200 to 80 > ms. Packet loss is zero. The fact that SIP->SIP works fine suggests the > issue isn't related to IP issues. > > I tried adding a jitter buffer, but that didn't make a difference. > > I've tried this sending just ULAW and G722 and allowing everything, but no > difference. The SDP that comes back from Paris doesn't list any audio > codecs and is: > > v=0 > o=default 1350406175 1350406175 IN IP4 10.10.22.246 > s=Asterisk PBX 10.7.1 > c=IN IP4 10.10.22.246 > t=0 0 > m=audio 32000 RTP/AVP 0 101 > a=sendrecv > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=maxptime:90 > m=video 0 RTP/AVP 31 34 34 98 99 104 > a=rtpmap:31 H261/90000 > a=rtpmap:34 H263/90000 > a=rtpmap:34 H263/90000 > a=rtpmap:98 h263-1998/90000 > a=rtpmap:99 H264/90000 > a=rtpmap:104 MP4V-ES/90000 > a=sendrecv > > Does anybody have any ideas as to what I should look at next? > > cat proc/interrupts? http://wiki.openvox.cn/index.php/Troubleshooting_of_PRI_cards Thanks, Steve T
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