[asterisk-users] BLF and Asterisk 1.6.0b2
Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy hints to phones? I'm not reporting this a s a bug because (although I have it working with Asterisk 1.4.17, the hardware involved is different. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF and Asterisk 1.6.0b2
Thomas Kenyon wrote: Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy hints to phones? I'm not reporting this a s a bug because (although I have it working with Asterisk 1.4.17, the hardware involved is different. What type of device are you subscribing to, is it another SIP phone? If so, what is the associated configuration in sip.conf? Do you have call-limit set to some value, or the combination of callcounter and busylevel? If so, what are they set to? (You must have these options set for it to work) -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF and Asterisk 1.6.0b2
Russell Bryant wrote: Thomas Kenyon wrote: Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy hints to phones? I'm not reporting this a s a bug because (although I have it working with Asterisk 1.4.17, the hardware involved is different. What type of device are you subscribing to, is it another SIP phone? If so, what is the associated configuration in sip.conf? Do you have call-limit set to some value, or the combination of callcounter and busylevel? If so, what are they set to? (You must have these options set for it to work) I have enough kit around to set the machine I'm testing 1.6.0b2 to use the same configuration as the working machines. I have got call-limits set, but it did occur to me that there's no reason asterisk would know that there is only one extension on SIP/peername. The stranger thing is, on the machine that it's all working on, there is a call-limit=4 set on every extension (from what I remember it prevented a bug that got fixed ages ago and I didn't get round to lowering it again). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
