G'day all, I've been googling myself silly looking for help on this one but have come up blank.
I have an AVM Fritz!Card PCI, and I'm using chan_capi v 0.3.1 with * from CVS-HEAD-05/08/04-22:48:00. I can start * and make and receive calls on ISDN fine but after a few hours of * uptime, on any ISDN call I make or receive from my SIP handsets (7960 or ATA-186) I get bad audio: on the ATA-186 I get either no audio incoming from ISDN at all (I get a scratch of sound every 2-2.5 seconds or so, but otherwise nothing) or robotic stuttering sound. On the 7960 the problem is just the no-audio-at-all. On either, outgoing audio (sound going out to ISDN) is fine. Originally I thought the problem was *all* ISDN calls, so I was debugging chan_capi. I have a couple of analogue handsets attached to a TDM400P though, and tried a call through that: perfect. I can receive a call on the 7960 receiving no audio, transfer it to the TDM phone and get perfect audio, transfer it to the ATA and get bad/no audio. The last idea I came up with was a WAG to do with codecs - my phones all used ulaw and my ISDN will be using alaw; even though * would have transcoded I thought it would work better if I switched the phones to alaw... As expected, no difference (I really didn't expect it to make a difference). To me, it looks like a variation of the SIP RTP timestamp problem (yes, my 7960 is at 6.3 code), but the problem exists on the ATA-186 too and I don't have any other issues with that (not even SIP-IAX, where the 7960 is really bad). Also strange is that there is no problem at all on my X-Lite softphone; I did the same thing transferring a call back and forth from X-Lite to the 7960 as I did with the analogue handset above. I guess this is going to come down to the old "crawl over the ethereal dump with a microscope" kind of thing, but I'm hoping that someone out there has solved this issue before. I know there's plenty of chan_capi users with ISDN BRI via a Fritz!Card, maybe somebody has had a similar problem and can help? Please? Happy to try and produce traces etc for diagnosis. Cheers, Vic Cross _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
