Hi as suggested in the group I have downloaded the Ast@home installed one of my PC for testing and made 2 extentions to test iam able to talk each other now i have setup one Trunk and made Out going when ever i call to out side i get a voice tone saying that all trunks are busy how can i resolve this problem ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051201/a03c6eba/attachment.htm
Giovanni Miano
2005-Dec-01 02:57 UTC
[Asterisk-Users] unable to make calls out using ast@home
Have u availability tranks ? 2005/12/1, ram <talk2ram@gmail.com>:> Hi > > as suggested in the group > I have downloaded the Ast@home > installed one of my PC for testing > > and made 2 extentions to test > iam able to talk each other > > now i have setup one Trunk > and made Out going > > when ever i call to out side > > i get a voice tone saying that all trunks are busy > > how can i resolve this problem > > ram > _______________________________________________ > --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 > > >-- Giovanni Miano
yes i got new account from provider and i have registered no one using, iam using that account for testing ram On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote:> > Have u availability tranks ? > > 2005/12/1, ram <talk2ram@gmail.com>: > > Hi > > > > as suggested in the group > > I have downloaded the Ast@home > > installed one of my PC for testing > > > > and made 2 extentions to test > > iam able to talk each other > > > > now i have setup one Trunk > > and made Out going > > > > when ever i call to out side > > > > i get a voice tone saying that all trunks are busy > > > > how can i resolve this problem > > > > ram > > _______________________________________________ > > --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 > > > > > > > > > -- > Giovanni Miano > _______________________________________________ > --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/20051201/5321e8c1/attachment.htm
Giovanni Miano
2005-Dec-01 03:11 UTC
[Asterisk-Users] unable to make calls out using ast@home
type in console: sip show registry and verify status of your trunk 2005/12/1, ram <talk2ram@gmail.com>:> yes > > i got new account from provider > and i have registered > > no one using, iam using that account for testing > > ram > > > On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote: > > > > Have u availability tranks ? > > > > 2005/12/1, ram <talk2ram@gmail.com >: > > > Hi > > > > > > as suggested in the group > > > I have downloaded the Ast@home > > > installed one of my PC for testing > > > > > > and made 2 extentions to test > > > iam able to talk each other > > > > > > now i have setup one Trunk > > > and made Out going > > > > > > when ever i call to out side > > > > > > i get a voice tone saying that all trunks are busy > > > > > > how can i resolve this problem > > > > > > ram > > > _______________________________________________ > > > --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 > > > > > > > > > > > > > > > -- > > Giovanni Miano > > _______________________________________________ > > --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 > > > > > _______________________________________________ > --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 > > >-- Giovanni Miano
Hi here is the results asterisk1*CLI> sip show registry Host Username Refresh State x.x.x..2:5060 xxxxxxxxxx 105 Registered i have edited the orginals ram On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote:> > type in console: sip show registry > and verify status of your trunk > > 2005/12/1, ram <talk2ram@gmail.com>: > > yes > > > > i got new account from provider > > and i have registered > > > > no one using, iam using that account for testing > > > > ram > > > > > > On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote: > > > > > > Have u availability tranks ? > > > > > > 2005/12/1, ram <talk2ram@gmail.com >: > > > > Hi > > > > > > > > as suggested in the group > > > > I have downloaded the Ast@home > > > > installed one of my PC for testing > > > > > > > > and made 2 extentions to test > > > > iam able to talk each other > > > > > > > > now i have setup one Trunk > > > > and made Out going > > > > > > > > when ever i call to out side > > > > > > > > i get a voice tone saying that all trunks are busy > > > > > > > > how can i resolve this problem > > > > > > > > ram > > > > _______________________________________________ > > > > --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 > > > > > > > > > > > > > > > > > > > > > -- > > > Giovanni Miano > > > _______________________________________________ > > > --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 > > > > > > > > > _______________________________________________ > > --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 > > > > > > > > > -- > Giovanni Miano > _______________________________________________ > --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/20051201/3bc1c907/attachment.htm
Giovanni Miano
2005-Dec-01 03:30 UTC
[Asterisk-Users] unable to make calls out using ast@home
pastme context for outgoing 2005/12/1, ram <talk2ram@gmail.com>:> Hi > > here is the results > > asterisk1*CLI> sip show registry > Host Username Refresh State > x.x.x..2:5060 xxxxxxxxxx 105 Registered > > > i have edited the orginals > > ram > > > > On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote: > > > > type in console: sip show registry > > and verify status of your trunk > > > > 2005/12/1, ram < talk2ram@gmail.com>: > > > yes > > > > > > i got new account from provider > > > and i have registered > > > > > > no one using, iam using that account for testing > > > > > > ram > > > > > > > > > On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote: > > > > > > > > Have u availability tranks ? > > > > > > > > 2005/12/1, ram < talk2ram@gmail.com >: > > > > > Hi > > > > > > > > > > as suggested in the group > > > > > I have downloaded the Ast@home > > > > > installed one of my PC for testing > > > > > > > > > > and made 2 extentions to test > > > > > iam able to talk each other > > > > > > > > > > now i have setup one Trunk > > > > > and made Out going > > > > > > > > > > when ever i call to out side > > > > > > > > > > i get a voice tone saying that all trunks are busy > > > > > > > > > > how can i resolve this problem > > > > > > > > > > ram > > > > > _______________________________________________ > > > > > --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 > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > Giovanni Miano > > > > _______________________________________________ > > > > --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 > > > > > > > > > > > > > _______________________________________________ > > > --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 > > > > > > > > > > > > > > > -- > > Giovanni Miano > > _______________________________________________ > > --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 > > > > > _______________________________________________ > --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 > > >-- Giovanni Miano
hi iam not sure is the right answer iam posting let me know is this correct or not Sip.conf ------------ [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown sip_additional.conf register=myaccount:mypassword@sipprovider [tel] username=myaccount type=peer secret=mysecret host=sipprovider IP ram On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote:> > pastme context for outgoing > > 2005/12/1, ram <talk2ram@gmail.com>: > > Hi > > > > here is the results > > > > asterisk1*CLI> sip show registry > > Host Username Refresh State > > x.x.x..2:5060 xxxxxxxxxx 105 Registered > > > > > > i have edited the orginals > > > > ram > > > > > > > > On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote: > > > > > > type in console: sip show registry > > > and verify status of your trunk > > > > > > 2005/12/1, ram < talk2ram@gmail.com>: > > > > yes > > > > > > > > i got new account from provider > > > > and i have registered > > > > > > > > no one using, iam using that account for testing > > > > > > > > ram > > > > > > > > > > > > On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote: > > > > > > > > > > Have u availability tranks ? > > > > > > > > > > 2005/12/1, ram < talk2ram@gmail.com >: > > > > > > Hi > > > > > > > > > > > > as suggested in the group > > > > > > I have downloaded the Ast@home > > > > > > installed one of my PC for testing > > > > > > > > > > > > and made 2 extentions to test > > > > > > iam able to talk each other > > > > > > > > > > > > now i have setup one Trunk > > > > > > and made Out going > > > > > > > > > > > > when ever i call to out side > > > > > > > > > > > > i get a voice tone saying that all trunks are busy > > > > > > > > > > > > how can i resolve this problem > > > > > > > > > > > > ram > > > > > > _______________________________________________ > > > > > > --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 > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > > Giovanni Miano > > > > > _______________________________________________ > > > > > --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 > > > > > > > > > > > > > > > > > _______________________________________________ > > > > --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 > > > > > > > > > > > > > > > > > > > > > -- > > > Giovanni Miano > > > _______________________________________________ > > > --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 > > > > > > > > > _______________________________________________ > > --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 > > > > > > > > > -- > Giovanni Miano > _______________________________________________ > --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/20051201/b9903fab/attachment.htm
Giovanni Miano
2005-Dec-01 03:45 UTC
[Asterisk-Users] unable to make calls out using ast@home
Context in extensions.conf 2005/12/1, ram <talk2ram@gmail.com>:> hi > > iam not sure is the right answer iam posting > let me know is this correct or not > > > Sip.conf > ------------ > > > [general] > > port = 5060 ; Port to bind to (SIP is 5060) > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) > disallow=all > allow=ulaw > allow=alaw > context = from-sip-external ; Send unknown SIP callers to this context > callerid = Unknown > > > > > sip_additional.conf > > > > register=myaccount:mypassword@sipprovider > > [tel] > username=myaccount > type=peer > secret=mysecret > host=sipprovider IP > > ram > > > > On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote: > > > > pastme context for outgoing > > > > 2005/12/1, ram <talk2ram@gmail.com >: > > > Hi > > > > > > here is the results > > > > > > asterisk1*CLI> sip show registry > > > Host Username Refresh > State > > > x.x.x..2:5060 xxxxxxxxxx 105 Registered > > > > > > > > > i have edited the orginals > > > > > > ram > > > > > > > > > > > > On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote: > > > > > > > > type in console: sip show registry > > > > and verify status of your trunk > > > > > > > > 2005/12/1, ram < talk2ram@gmail.com>: > > > > > yes > > > > > > > > > > i got new account from provider > > > > > and i have registered > > > > > > > > > > no one using, iam using that account for testing > > > > > > > > > > ram > > > > > > > > > > > > > > > On 12/1/05, Giovanni Miano <giomiano@gmail.com> wrote: > > > > > > > > > > > > Have u availability tranks ? > > > > > > > > > > > > 2005/12/1, ram < talk2ram@gmail.com >: > > > > > > > Hi > > > > > > > > > > > > > > as suggested in the group > > > > > > > I have downloaded the Ast@home > > > > > > > installed one of my PC for testing > > > > > > > > > > > > > > and made 2 extentions to test > > > > > > > iam able to talk each other > > > > > > > > > > > > > > now i have setup one Trunk > > > > > > > and made Out going > > > > > > > > > > > > > > when ever i call to out side > > > > > > > > > > > > > > i get a voice tone saying that all trunks are busy > > > > > > > > > > > > > > how can i resolve this problem > > > > > > > > > > > > > > ram > > > > > > > _______________________________________________ > > > > > > > --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 > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > > > Giovanni Miano > > > > > > _______________________________________________ > > > > > > --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 > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > --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 > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > Giovanni Miano > > > > _______________________________________________ > > > > --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 > > > > > > > > > > > > > _______________________________________________ > > > --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 > > > > > > > > > > > > > > > -- > > Giovanni Miano > > _______________________________________________ > > --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 > > > > > _______________________________________________ > --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 > > >-- Giovanni Miano
Peter Hoppe
2005-Dec-28 06:02 UTC
[Asterisk-Users] unable to make calls out using ast@home
I am setting up a phone system using Asterisk@home, version 1.5. It runs Asterisk 1.0.9 built by root@asterisk1.local on a i686 running Linux (Asterisk info). I had some bigger problems: In AmpPortal / Setup/ Extensions: When I added new SIP devices and then looked at the resulting sip.conf I saw that the file got messed up - per extension settings were duplicated. As result the SIP devices didn't register anymore. I then hand edited my sip.conf and devices did register successfully. I then added estensions, but when I tried to initiate phone calls, no phone rang. So I hand edited extensions.conf as well, and lo-and-behold it worked! Since I have some tighter deadline I decided that it wasn't woth trying to use the AMP-portal way of things and simply scrapped the config files which were offered and to use hand edited files instead. System works very well now (except some features I still have to implement). Despite of all this I am NOT disappointed about a@home - I think it's a great software package, and I am very grateful that there are people who take the trouble of setting all that up and to offer it in such an easy-to-install package. The most likely reason that it didn't work is probably my own ignorance. There would probably be thousands of people who successfully used a@home as well. I just didn't have time to fiddle with it. But the system works now fine with hand edited files. Peter