Hi, I've got asterisk working as a PSTN gateway between two sites. The IP connection is pretty good and bandwidth exceeds 640Kbps all the time. I'm using X100P hardware on 300Mhz and 333MHz systems. One of them has a sound card, but the slower system does not have a sound card. Both are connected to PBXes and I have busydetect=yes on one end because the PBX doesn't drop the voltages at disconnect. The other has a more normal ATA for the analog port.
I've searched the FAQ, mailing list archives, and documentation and I don't see this matter addressed. I'd appreciate any help you all can offer. My problem is that the calls drop suddenly while the call is in progress, at random times. Sometimes the call proceeds for several minutes before dropping, and sometimes it drops within seconds. I have bandwidth=low in iax.conf on both ends. I don't even know what else to look at or what to try to change, to make this problem go away (or to be less apparent). What I want is for the IAX systems on each end to be a little more "forgiving" of IP latency/jitter etc. before abruptly disconnecting/dropping the call. Please let me know what sorts of things I can try to do to make the VoIP calls more reliable between these gateways. Thank you! -- Jim Ockers ([EMAIL PROTECTED]) Contact info: please see http://www.ockers.net/ Fight Spam! Join CAUCE (Coalition Against Unsolicited Commercial Email) at http://www.cauce.org/ . _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
