Kubat, Philip <> scribbled on Wednesday, June 23, 2004 2:32 PM: > I have a SIP connection to Broadvoice and sometimes when I make > outgoing calls from a SIP ATA-188 (could be the same number) (the > ATA-188, is currently the only extension), there is no audio passed > for 5-10 secs. I have set all the codec the same to 711u and also > ensured canreinvite is set to no. > > > > Any suggestions? Places to look for?
Philip, I'm not quite sure what you are asking here. Are you connecting directly to Broadvoice with your ATA-188? If so, you would need to contact Broadvoice support in this case. If you are connecting to Broadvoice with Asterisk, then to Asterisk with your ATA-188, you probably have either a configuration problem with your .conf files or a firewall issue that is taking a long time to sort out. Take a look at the Wiki page Asterisk Settings Broadvoice. (http://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20Broa dvoice) It is based off of my configuration which gives me no delay at all. Since I am behind a NAT, I also forwarded UDP ports 5060 and the port range listed in rtp.conf to my Asterisk box as well. Something else that may be causing you grief is the dial plan in your ATA-188. I am not familiar with that model as I am currently using a Sipura, but you may have a case where you are dialing, and hitting the digit timeout before the SIP initiation takes place. Try dialing your number then pressing the Pound (#) key and see if the dialing is immediate. If it is, check the configuration of your ATA and see if you can either shorten the timeout period or create a pattern match that will pick up the shorter dial string. Here is what I would try (in this order) in your shoes: 1. Try dialing with a # key and see if that fixes it. Modify ATA config if it does. 2. Modify your firewall rules to forward the above mentioned ports to your Asterisk box. 3. Compare what you have in your sip.conf with the Wiki. If you have different configuration lines, modify yours to more closely match the working example. 4. If none of the above ideas work, reply with the relevant sections of your sip.conf and extensions.conf to this thread and somebody may be able to see something that is not correct. Good luck, Jeremy _______________________________________________ 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
