5 posts / 0 new
Last post
dozment
dozment's picture
Joined: 2007/11/20
Points: -10

BANG!!!!!

That was the sound of me shooting myself in the foot. I did an asterisk upgrade from 1.4.0 to 1.4.18 last night, and I tested almost everything. The one thing I missed was an IVR that plays a brief intro message and then drops to the main IVR. There is no prompt (or time) to enter a keystroke in the first short message. It's just there to give the customer an easy way to add an announcement to their greeting without having to re-record the whole main recording.

Well, I found out this morning that it was playing the first recording and then failing. Scratched my head about it for some time. Then, I remembered Alex telling me some time back to set autofallthrough=no in extensions.conf. At some point along the way the default for authfallthrough becamse yes. That caused me a lot of grief today, so I thought I'd pass it along.

eeman
eeman's picture
Joined: 2007/11/06
Points: 290

wow i cant believe you ran 1.4.0 for so long and didnt have people calling for your public execution. :-) the bugs in 1.4.0 - 1.4.10 were rather severe including some absolutely horrid DTMF recognition.

Erik Smith
dCAP
Thirdlane/Asterisk Support available
esmith.bgnv@gmail.com

dozment
dozment's picture
Joined: 2007/11/20
Points: -10

Well, I guess I'm a glutton for punishment. I'm glad to hear you say it gets better after 1.4.0. I've been through DTMF hell, and I hope that's over. I've also recently had a problem where audio just kind of stops on inbound and outbound calls for about 10 seconds. The calls don't disconnect. If you hang on long enough it goes back to normal. I saw a reference to on astrisk.org from 8/2007, but there was never a comment on what might cause it. I'm hoping that will be better with 1.4.18 as well.

eeman
eeman's picture
Joined: 2007/11/06
Points: 290

I've run into jitter buffer issues that create one-way audio. I have disabled all iax and sip jitter buffers and now just use the zaptel jitter buffer setting. they say jitter buffers should only be used on ip end points so hopefully the phones' jitter buffers will buffer the receiving stream.

Erik Smith
dCAP
Thirdlane/Asterisk Support available
esmith.bgnv@gmail.com

dozment
dozment's picture
Joined: 2007/11/20
Points: -10

Thanks for the tip. I've got JB disabled in sip.conf.