Helloi am trying to setup sipgate gatewayi can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line– Called 01179248615@sipgate[Sep 18 13:58:30] NOTICE: chan_sip.c:17885handle_response_invi..
I have only seen this problem when using SIPgate SIP trunks which actually register. If the ADSL connection goes down that the sip trunk uses, the sip phones registered locally become unreachable. This happens on any 1.6.x or 1.8 version of aster..
a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the lo..
I am using the voicemail module of asterisk. When I did some test calls from my mobile phone, sometimes the beginning of the prompt was missing, e.g. instead of something like number 12345 not available I was only hearing 345 not available. Verbose le..
I configured asterisk in sip.conf like that: ===== register => username:firstname.lastname@example.org:5060/number [sipgate-out] port=5060 type=friend insecure=invite nat=yes username=username fromuser=username fromdomain=sipgate.de secret=secret host=sipgate..
list, I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to sipgate.co.uk as a sip agent/client (with register => statement in sip.conf). If I restrict the number of ports used in rtp.conf (to 10000-10005 for example) – will that affect ..