Charlie Grosvenor
2006-Nov-15 13:40 UTC
[asterisk-users] PortSip and Astericks new install
I have just installed Asterisk and installed the sample configuration files. Asterisks appears to be working and I have added a SIP client: [John] type=friend secret=test host=dynamic allow=all I have been trying to dial the demo number 500 when using PortSip, Asterisks answers the phone but PortSip gives me the error: "Call failed: codec not accepted 488". I have tried changing the enabled codecs in PortSip but this makes no difference. I have also tried various other SoftPhone but none of them seem to work. Anybody know what I have missed / doing wrong? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061115/6c54761a/attachment.htm
try : [John] type=friend secret=test host=dynamic disallow=all allow =gsm&ilbc&ulaw&alaw Also try other sip phone slike sjphone just to make sure there is no prob . On 16/11/06, Charlie Grosvenor <Charlie@cgrosvenor.co.uk> wrote:> > I have just installed Asterisk and installed the sample configuration > files. Asterisks appears to be working and I have added a SIP client: > > > > [John] > > type=friend > > secret=test > > host=dynamic > > allow=all > > > > I have been trying to dial the demo number 500 when using PortSip, > Asterisks answers the phone but PortSip gives me the error: > > > > "Call failed: codec not accepted 488". > > > > I have tried changing the enabled codecs in PortSip but this makes no > difference. I have also tried various other SoftPhone but none of them seem > to work. > > > > Anybody know what I have missed / doing wrong? > > > > Thanks > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061115/84d1a99f/attachment.htm
Charlie Grosvenor wrote:> > > > [John] > > type=friend > > secret=test > > host=dynamic > > allow=all >Try: disallow=all allow=alaw allow=ulaw allow=gsm Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
Charlie Grosvenor
2006-Nov-15 15:36 UTC
[asterisk-users] PortSip and Astericks new install
Thanks for your reply, I have tried what you have suggested and get the same issue. I have also tried express talk which connects then say: Initiated sip call to 500 Call has disconnected. Any other ideas? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Doug Lytle Sent: 15 November 2006 21:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PortSip and Astericks new install Charlie Grosvenor wrote:> > > > [John] > > type=friend > > secret=test > > host=dynamic > > allow=all >Try: disallow=all allow=alaw allow=ulaw allow=gsm Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users