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Hi i got lots of this from the asterisk console what does this mean? format_wav.c:247 update_header: Unable to find our position asterisk console: Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[2812]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[4430]: format_wav.c:247 update_header: U...

Does anyone know what this means "Warning [65542]: chan_sip:c:501 retrans_plct: Maximum retries exceeded on call 7438737dc873850@172.16.0.52 for seqno102 (Non-critical Request. 172.16.0.52 is the Asterisk Server I'm guessing that I have something miss configured just not sure what it is. If you need more info just ask.

...e CLI. Dell PE 1600 with Sangoma A200. pbtpbx*CLI> show version Asterisk 1.2.14 built by root @ pbtpbx.local on a i686 running Linux on 2007-01-13 18:31:56 UTC Asterisk Queue Logger restarted Rotated Logs Per SIGXFSZ (Exceeded file size limit) Feb 5 08:43:00 WARNING[20103]: format_wav.c:247 update_header: Unable to find our position == Parsing '/etc/asterisk/logger.conf': Found -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com -- This message has been scanned for...

...nitor(test.wav) exten => 22,2,Dial(SIP/blabla@blabla.com) No problems at all if I record to a file, but then I made test.wav a fifo, and had oggenc read it, then pipe it to oggfwd, etc... This does work, but generates a pile of warnings in the asterisk console: WARNING[16235]: format_wav.c:247 update_header: Unable to find our position I can shut off the warnings, of course, using logger.conf, but it still seems kind of messy. I also need to start the commands reading that pipe manually after initiating the call. Can anyone suggest a better way to do this? I figure an AGI script could probably sta...

Hi all, I’m trying to rewrite Diversion header when call forwarding is done on the phone. The phone sends "302 Moved Temporarily" response and sets Diversion header to a local number, but before Asterisk sends this call towards TSP provider I need to change Diversion header to a full PSTN number. I am using PJSIP_HEADER in a pre-dial handler (configuration is below). On the same