[asterisk-users] Interpbx connection
khalid touati
khalidtouati at gmail.com
Wed Apr 21 13:55:14 CDT 2010
Steve,
You're completely right!! it seems like my colleague gave me a wrong info
(probably a firewall issue), i was also curious (before i read your
response) so i tried this in my network and really it has nothing to do with
call setup or peer authentication, sorry for the wrong info Guys!
2010/4/19 Steve Edwards <asterisk.org at sedwards.com>
> Un-top-posting...
>
> > 2010/4/14 khalid touati <khalidtouati at gmail.com>
>
> > i've connecting two pbx server successfully for several times using
> the following config:
> >
> > register => USPBX:mypass at 122.11.176.35<USPBX%3Amypass at 122.11.176.35>
> >
> > [PBX1]
> > type=friend
> > host=122.11.176.35
> > trunk=yes
> > sercret=mypass
> > context=external
> > deny=0.0.0.0/0.0.0.0
> > permit=122.11.176.35/255.255.255.240
> > insecure=very
> > allow=all
> > nat=yes
> > qualify=yes
> > canreinvite=no
> >
> > in the other and it's the analog.
> >
> > but now i can only dial from one end, and the other en d is giving
> me this error.
> >
> > Apr 14 16:44:21 ERROR[26502]: chan_sip.c:6659 register_verify: Peer
> 'PBX1' is trying to register, but not
> > configured as host=dynamic
> >
> > when dialing a fast busy signal and it sauys in the CLI:
> CONGESTION. any help please!!!
> >
> > --
> > Abdullah
>
> On Mon, 19 Apr 2010, khalid touati wrote:
>
> > for people's future references: we found out that the option in DIAL
> > application in the extensions.conf has to be the same from both side,
> > the issue was India server was using "tr" while US server was using
> > "TWw" so we made them both using "tr" and that solved the issue, i guess
> > if one side is set to "trTWw" that would work regardless of the other
> > side but didn't try though. have a headeache-free experience with
> > asterisk "the future of telephony" :)!
>
> Dial() options don't have any relationship to registration failures --
> they happen at different times.
>
> Registration failures may cause dial() failures.
>
> I don't understand the relationship between ringing, transfer and
> recording options and dial() returning congestion. I'd suggest
> investigating exactly which combination is causing congestion before
> concluding it is unrelated to the registration failure.
>
> --
> Thanks in advance,
> -------------------------------------------------------------------------
> Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Abdullah
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100421/a544b2c7/attachment.htm
More information about the asterisk-users
mailing list