Jerry Geis
2006-Sep-14 05:13 UTC
[asterisk-users] asterisk server to server using sip question
I have 2 asterisk servers. I am trying to connect them with SIP and getting an error. My first box I define sip.conf as: [devcentos64_to_bt610tMM] type=friend username=devcentos64_to_bt610tMM secret=password disallow=all allow=ulaw allow=alaw allow=gsm host=192.168.1.159 context=default my second box I define sip.conf as: [devcentos64_to_bt610tMM] type=friend username=devcentos64_to_bt610tMM secret=password disallow=all allow=ulaw allow=alaw allow=gsm host=192.168.1.10 context=default So Box2 points to Box1 and Box1 points to Box2 by the host= fields. I am getting the following error: -- Attempting call on SIP/devcentos64_to_bt610tmm/1041 for smvoice_callprogress@smvoice-dialout:1 (Retry 1) Sep 14 08:10:52 WARNING[2512]: chan_sip.c:9715 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Jerry Geis 204" <sip:3175661677@192.168.1.10>;tag=as07330b38' > Channel SIP/devcentos64_to_bt610tmm-007afe00 was never answered. Sep 14 08:10:52 WARNING[4639]: cdr.c:550 ast_cdr_disposition: Cause not handled Why is that??? My passwords match. I am using asterisk.1.2.11 Or what is the correct way to connect asterisk SIP server to asterisk SIP server. Jerry
I am a new member and I got this error message: rmv aiu stoped data base error tks ----- Original Message ----- From: "Jerry Geis" <geisj@pagestation.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, September 14, 2006 9:13 AM Subject: [asterisk-users] asterisk server to server using sip question>I have 2 asterisk servers. I am trying to connect them with SIP and > getting an error. > My first box I define sip.conf as: > > [devcentos64_to_bt610tMM] > type=friend > username=devcentos64_to_bt610tMM > secret=password > disallow=all > allow=ulaw > allow=alaw > allow=gsm > host=192.168.1.159 > context=default > > my second box I define sip.conf as: > [devcentos64_to_bt610tMM] > type=friend > username=devcentos64_to_bt610tMM > secret=password > disallow=all > allow=ulaw > allow=alaw > allow=gsm > host=192.168.1.10 > context=default > > So Box2 points to Box1 and Box1 points to Box2 by the host= fields. > > I am getting the following error: > -- Attempting call on SIP/devcentos64_to_bt610tmm/1041 for > smvoice_callprogress@smvoice-dialout:1 (Retry 1) > Sep 14 08:10:52 WARNING[2512]: chan_sip.c:9715 handle_response_invite: > Forbidden - wrong password on authentication for INVITE to '"Jerry Geis > 204" <sip:3175661677@192.168.1.10>;tag=as07330b38' > > Channel SIP/devcentos64_to_bt610tmm-007afe00 was never answered. > Sep 14 08:10:52 WARNING[4639]: cdr.c:550 ast_cdr_disposition: Cause not > handled > > > Why is that??? My passwords match. I am using asterisk.1.2.11 > Or what is the correct way to connect asterisk SIP server to asterisk > SIP server. > > Jerry > > > _______________________________________________ > --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 > >
Luiz Miguel wrote:> I am a new member and I got this error message: > > rmv aiu stoped data base errorthis sounds like a rather big problem> tks
Luiz Miguel wrote:> I am a new member and I got this error message: > > rmv aiu stoped data base error >google for it... it found a few interesting references, among which: http://www.textfiles.com/magazines/TOT/tot-o6.txt
Yes . I know. Do you have the ODBE forms related page 10.16 and 10.17. These pages are about AIU ( EUAIU and APAIU) Tks ----- Original Message ----- From: "Rapha?l Jacquot" <sxpert@esitcom.org> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Thursday, September 14, 2006 10:11 AM Subject: Re: [asterisk-users] is there anyone working with 5ESS?> Luiz Miguel wrote: >> I am a new member and I got this error message: >> >> rmv aiu stoped data base error > > this sounds like a rather big problem > >> tks > > _______________________________________________ > --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 > >
Steven Totaro
2006-Sep-14 08:38 UTC
[asterisk-users] asterisk server to server using sip question
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Jerry Geis > Sent: Thursday, September 14, 2006 8:14 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] asterisk server to server using sip question > > I have 2 asterisk servers. I am trying to connect them with SIP and > getting an error. > My first box I define sip.conf as: > > [devcentos64_to_bt610tMM] > type=friend > username=devcentos64_to_bt610tMM > secret=password > disallow=all > allow=ulaw > allow=alaw > allow=gsm > host=192.168.1.159 > context=default > > my second box I define sip.conf as: > [devcentos64_to_bt610tMM] > type=friend > username=devcentos64_to_bt610tMM > secret=password > disallow=all > allow=ulaw > allow=alaw > allow=gsm > host=192.168.1.10 > context=default > > So Box2 points to Box1 and Box1 points to Box2 by the host= fields. > > I am getting the following error: > -- Attempting call on SIP/devcentos64_to_bt610tmm/1041 for > smvoice_callprogress@smvoice-dialout:1 (Retry 1) > Sep 14 08:10:52 WARNING[2512]: chan_sip.c:9715 handle_response_invite: > Forbidden - wrong password on authentication for INVITE to '"JerryGeis> 204" <sip:3175661677@192.168.1.10>;tag=as07330b38' > > Channel SIP/devcentos64_to_bt610tmm-007afe00 was neveranswered.> Sep 14 08:10:52 WARNING[4639]: cdr.c:550 ast_cdr_disposition: Causenot> handled > > > Why is that??? My passwords match. I am using asterisk.1.2.11 > Or what is the correct way to connect asterisk SIP server to asterisk > SIP server. > > Jerry > >Do you have other peers on the same boxes pointing to each other?
Jerry Geis
2006-Sep-14 10:25 UTC
[asterisk-users] asterisk server to server using sip question
Steve, Yes Box1 does have multiple other peers. Do you know of something to try? Jerry> / -----Original Message-----/>/ From: asterisk-users-bounces at lists.digium.com <http://lists.digium.com/mailman/listinfo/asterisk-users> [mailto:asterisk-users- />/ bounces at lists.digium.com <http://lists.digium.com/mailman/listinfo/asterisk-users>] On Behalf Of Jerry Geis />/ Sent: Thursday, September 14, 2006 8:14 AM />/ To: asterisk-users at lists.digium.com <http://lists.digium.com/mailman/listinfo/asterisk-users> />/ Subject: [asterisk-users] asterisk server to server using sip question />/ />/ I have 2 asterisk servers. I am trying to connect them with SIP and />/ getting an error. />/ My first box I define sip.conf as: />/ />/ [devcentos64_to_bt610tMM] />/ type=friend />/ username=devcentos64_to_bt610tMM />/ secret=password />/ disallow=all />/ allow=ulaw />/ allow=alaw />/ allow=gsm />/ host=192.168.1.159 />/ context=default />/ />/ my second box I define sip.conf as: />/ [devcentos64_to_bt610tMM] />/ type=friend />/ username=devcentos64_to_bt610tMM />/ secret=password />/ disallow=all />/ allow=ulaw />/ allow=alaw />/ allow=gsm />/ host=192.168.1.10 />/ context=default />/ />/ So Box2 points to Box1 and Box1 points to Box2 by the host= fields. />/ />/ I am getting the following error: />/ -- Attempting call on SIP/devcentos64_to_bt610tmm/1041 for />/ smvoice_callprogress at smvoice-dialout <http://lists.digium.com/mailman/listinfo/asterisk-users>:1 (Retry 1) />/ Sep 14 08:10:52 WARNING[2512]: chan_sip.c:9715 handle_response_invite: />/ Forbidden - wrong password on authentication for INVITE to '"Jerry /Geis> / 204" <sip:3175661677 at 192.168.1.10 > <http://lists.digium.com/mailman/listinfo/asterisk-users>>;tag=as07330b38' >/>/ > Channel SIP/devcentos64_to_bt610tmm-007afe00 was never /answered.> / Sep 14 08:10:52 WARNING[4639]: cdr.c:550 ast_cdr_disposition: Cause/not> / handled/>/ />/ />/ Why is that??? My passwords match. I am using asterisk.1.2.11 />/ Or what is the correct way to connect asterisk SIP server to asterisk />/ SIP server. />/ />/ Jerry />/ />/ / Do you have other peers on the same boxes pointing to each other?