[asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops
Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops
do you use the qualify=yes option for your endpoints? y. Peter Childs schrieb: Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops
2010/1/26 Yves Arikoglu yves...@gmx.de: do you use the qualify=yes No, If I do it does not work at all. I've found if I set defaultexpiry to 30 it works fine. and was infact working for 30 seconds every two minutes before, It looks like sipgate.co.uk are expiring there registry attempts very quickly. Peter option for your endpoints? y. Peter Childs schrieb: Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops
2010/1/26 Peter Childs pchi...@bcs.org: 2010/1/26 Yves Arikoglu yves...@gmx.de: do you use the qualify=yes No, If I do it does not work at all. I've found if I set defaultexpiry to 30 it works fine. and was infact working for 30 seconds every two minutes before, It looks like sipgate.co.uk are expiring there registry attempts very quickly. However I'm not totally sure this fixes the whole problem, as it still only works sometimes. Its just its works more often now than it did before. Peter. Peter option for your endpoints? y. Peter Childs schrieb: Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
