Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well.
However, I noticed after the upgrade, when dialing into an IVR/voicemail, the first part of every audio file that is played gets cut off. This happens regardless of encoding of the file (ulaw/gsm) and regardless of the incoming codec. However when using Echo() both tones & voice are flawlessly echoed back to me, as are the Packet2Packet bridging calls connected to remote phones. I tested this issue with 3 other providers (Link2VoIP/Babytel/Junction Networks) and I'm not experiencing this issue with them, despite having identical peer configurations across for all 4. Though with Teliax I'm using SIP, I did try to use IAX2 for the heck of it and the same problem seems to exists, so it's not specific to SIP. Additionally, I tried changing Teliax proxies just for the heck of it and that made no difference. --- Example of what I see and then hear... --- -- <SIP/teliax-00000000> Playing 'vm-login.ulaw' (language 'en') -- <SIP/teliax-00000000> Playing 'vm-password.ulaw' (language 'en') -- <SIP/teliax-00000000> Playing 'vm-youhave.ulaw' (language 'en') -- <SIP/teliax-00000000> Playing 'vm-no.ulaw' (language 'en') -- <SIP/teliax-00000000> Playing 'vm-messages.ulaw' (language 'en') -- <SIP/teliax-00000000> Playing 'vm-opts.ulaw' (language 'en') -- <SIP/teliax-00000000> Playing 'vm-helpexit.ulaw' (language 'en') In this case, I'd hear "gin" "essages". The 'password', 'youhave', and 'no' prompts are actually so short you don't hear them at all. http://help.teliax.com/discussions/support/1924-asterisk-1620-rc6 --- I've contacted Teliax about this, but I suspect they're short handed due to the holiday weekend. Has anyone experienced this with 1.6.x & Teliax? And if so, what did you do to solve it (if anything)? I'd hate to revert, I spent a lot of time redoing my configs. :) Thanks in advance! Jeff _______________________________________________ -- 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