Russell Brown
2007-Nov-30 16:25 UTC
[asterisk-users] Simple Asterisk to Asterisk SIP Call Setup?
I have two Asterisk systems that can route to each other via a VPN with
firewalls disabled for testing purposes.
Each Server can see (tested via nmap) UDP port 5060 on the other.
So... I thought that I could simply use a Dial command in Server A's
config to place a SIP call to Server B... but it doesn't seem to work.
Server A (192.168.1.33) has:
exten => *136,1,Dial(SIP/90 at 10.10.111.13,30)
but whenever a user on Server A dials '*136' the call doesn't
complete
and the CLI shows:
Executing [*136 at from-sip:1] Dial("SIP/112-0071f650", "SIP/90
at 10.10.111.13|30") in new stack
-- Called 90 at 10.10.111.13
-- SIP/10.10.111.13-00793520 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
I can't see anything in Server B's logs from 192.168.1.33
What am I missing?
Any pointers to help me get this working?
--
Regards,
Russell
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| Russell Brown | MAIL: russell at lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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Vivek Shrivastava
2007-Nov-30 16:47 UTC
[asterisk-users] Simple Asterisk to Asterisk SIP Call Setup?
looks like something wrong with the dial plan in the extensions.conf.. i would recommend start debug on and see the content of "full" log may be that give some clue. Thanks, Vivek On 11/30/07, Russell Brown <russell at lls.lls.com> wrote:> > > I have two Asterisk systems that can route to each other via a VPN with > firewalls disabled for testing purposes. > > Each Server can see (tested via nmap) UDP port 5060 on the other. > > So... I thought that I could simply use a Dial command in Server A's > config to place a SIP call to Server B... but it doesn't seem to work. > > Server A (192.168.1.33) has: > > exten => *136,1,Dial(SIP/90 at 10.10.111.13,30) > > but whenever a user on Server A dials '*136' the call doesn't complete > and the CLI shows: > > Executing [*136 at from-sip:1] Dial("SIP/112-0071f650", " > SIP/90 at 10.10.111.13|30") in new stack > -- Called 90 at 10.10.111.13 > -- SIP/10.10.111.13-00793520 is circuit-busy > == Everyone is busy/congested at this time (1:0/1/0) > > I can't see anything in Server B's logs from 192.168.1.33 > > What am I missing? > > Any pointers to help me get this working? > > -- > Regards, > Russell > -------------------------------------------------------------------- > | Russell Brown | MAIL: russell at lls.com PHONE: 01780 471800 | > | Lady Lodge Systems | WWW Work: http://www.lls.com | > | Peterborough, England | WWW Play: http://www.ruffle.me.uk | > -------------------------------------------------------------------- > > _______________________________________________ > --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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071130/46a905ad/attachment.htm