Shaun Wingrin
2008-Dec-01 22:23 UTC
[asterisk-users] Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"
Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12 File: sip_custom.conf [VoipDirect777821] type=friend host=141.122.139 username=VoipDirect777821 secret=wsPiOov8830 accountcode=5260477782 amaflags=billing context=Incomming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes sip show peers shows both as registered. this is the error when try and place a call from Asterisk 1 to Asterisk 2: - Executing [582 at a1:1] Dial("Console/dsp", "SIP/VoipDirect777821|60|") in new stack -- Called VoipDirect777821 [Dec 1 23:20:21] WARNING[25399]: chan_sip.c:12334 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:asterisk at 141.122.139.16>;tag=as070b02e2' -- SIP/VoipDirect777821-0876c360 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [582 at a1:2] Hangup("Console/dsp", "") in new stack == Spawn extension (a1, 582, 2) exited non-zero on 'Console/dsp' << Hangup on console >> I get the same error even if I include this on Asterisk 1: register => VoipDirect777821:xxxxx at dfvvd.dyndns.org Please help.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081202/b1537d20/attachment.htm
Rafael Canchola
2008-Dec-01 22:31 UTC
[asterisk-users] Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"
You can use next parameter: Fromuser = VoipDirect777821 At 04:23 p.m. 01/12/2008, Shaun Wingrin wrote:>Please help. > >Asterisk 1: Sip.conf >[VoipDirect777821] >type=friend >host=dfvvd.dyndns.org >username=VoipDirect777821 >secret=xxxxxxxxxxxx >accountcode=5260477782 >amaflags=billing >context=Incoming >disallow=all >allow=g729 >;allow=alaw >;allow=ulaw >trunk=no >qualify=yes >qualifysmoothing=yes >nat=no >canreinvite=yes >dtmfmode=rfc2833 >;directrtpsetup=no >t38pt_udptl = yes > >Asterisk 2 sip.conf > GNU nano 1.3.12 File: sip_custom.conf > >[VoipDirect777821] >type=friend >host=141.122.139 >username=VoipDirect777821 >secret=wsPiOov8830 >accountcode=5260477782 >amaflags=billing >context=Incomming >disallow=all >allow=g729 >;allow=alaw >;allow=ulaw >trunk=no >qualify=yes >qualifysmoothing=yes >nat=no >canreinvite=yes >dtmfmode=rfc2833 >;directrtpsetup=no >t38pt_udptl = yes > >sip show peers shows both as registered. > >this is the error when try and place a call from Asterisk 1 to Asterisk 2: > >- Executing [582 at a1:1] Dial("Console/dsp", >"SIP/VoipDirect777821|60|") in new stack > -- Called VoipDirect777821 >[Dec 1 23:20:21] WARNING[25399]: chan_sip.c:12334 >handle_response_invite: Received response: "Forbidden" from >'"asterisk" <sip:asterisk at 141.122.139.16>;tag=as070b02e2' > -- SIP/VoipDirect777821-0876c360 is circuit-busy > == Everyone is busy/congested at this time (1:0/1/0) > -- Executing [582 at a1:2] Hangup("Console/dsp", "") in new stack > == Spawn extension (a1, 582, 2) exited non-zero on 'Console/dsp' > << Hangup on console >> > >I get the same error even if I include this on Asterisk 1: >register => VoipDirect777821:xxxxx at dfvvd.dyndns.org > >Please help.... > >_______________________________________________ >-- 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---------- RafaelCanchola Product Development Engineer, FonetGlobal Inc. rcm at fonetglobal.com http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 14 VoIP 523663801 d00d! cyberalph -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081201/0cd27347/attachment.htm
Danny Nicholas
2008-Dec-01 22:31 UTC
[asterisk-users] Inbound calls from Asterisk to Asterisk with SIP"Forbidden" from '"asterisk"
You shouldn't "open text" your password. Shouldn't IP on Asterisk 2 be 1.2.3.4? _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shaun Wingrin Sent: Monday, December 01, 2008 4:23 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Inbound calls from Asterisk to Asterisk with SIP"Forbidden" from '"asterisk" Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12 File: sip_custom.conf [VoipDirect777821] type=friend host=141.122.139 username=VoipDirect777821 secret=wsPiOov8830 accountcode=5260477782 amaflags=billing context=Incomming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes sip show peers shows both as registered. this is the error when try and place a call from Asterisk 1 to Asterisk 2: - Executing [582 at a1:1] Dial("Console/dsp", "SIP/VoipDirect777821|60|") in new stack -- Called VoipDirect777821 [Dec 1 23:20:21] WARNING[25399]: chan_sip.c:12334 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:asterisk at 141.122.139.16>;tag=as070b02e2' -- SIP/VoipDirect777821-0876c360 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [582 at a1:2] Hangup("Console/dsp", "") in new stack == Spawn extension (a1, 582, 2) exited non-zero on 'Console/dsp' << Hangup on console >> I get the same error even if I include this on Asterisk 1: register => VoipDirect777821:xxxxx at dfvvd.dyndns.org Please help.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081201/bf36b232/attachment.htm