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Showing articles from Asterisk tag

Low volume on voicemail

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…

Asterisk/Trixbox Trunk Sample Configurations

[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 & DTMF

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,…

I just get dead air on inbound call to my Asterisk server occasionally.

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.

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