If you have a SIP trunk and are using Asterisk for voicemail. we have found Asterisk records the volume quite low in its voicemail files. One approach is to use a post-voicemail command in the voicemail configuration, which can run Sox on the saved files and amplify them automatically. A recommended level for amp…
[vp.thinktel.ca] disallow=all allow=ulaw dtmfmode=rfc2833 fromdomain=vp.thinktel.ca host=159.18.161.67 username=XXXYYYZZZZ # your SIP authentication number secret=@@@@@@@@ # your SIP authentication password nat=yes # or nat=no if using publicly accessible IP insecure=invite,port type=fri…
Asterisk normally handles RFC2883 in a strange manner: From the phone: Audio Audio Audio DigitStart Duration Duration DigitEnd Audio Audio It waits until it sees 'DigitEnd' before it does anything, then it sends (in a tight for loop): DigitStart Start Start End End End (takes about 1ms normally) So,…
If your traces are showing the Thinktel switch responding with a 491 - Request Pending, you must set canreinvite=no to stop this. Asterisk does not always acknowledge this properly before any audio change is redone.