[Asterisk-Users] PRI to SIP Problem

[OTR Comm]

I'm pretty new to Asterisk, and have the CVS head from a week ago installed,
so I guess the fix is hidden someplace not so obvious.

I don't really understand what you mean by *Answer before Dial,* could you
explain that?  And I don't know how to use the *r* option for dialing on a
Cisco 7960.

Also, do you know where here is a searchable asterisk-users archive?

Thanks,
Murrah

----- Original Message ----- 
From: "Gary Reuter" <gary.reuter at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Thursday, October 27, 2005 6:27 PM
Subject: Re: [Asterisk-Users] PRI to SIP Problem


Yes, many people have had this problem.
Check the mailing list archives... I think the newest code has the fix.
Workaround for older versions is to Answer before Dial, but you may still
need the 'r' option to Dial as ringing may stop for the caller after about
10 seconds.

On 10/27/05, OTR Comm <otrcomm at isp-systems.net> wrote:
>
> Hello all,
>
> I have a problem calling into asterisk on a PRI going out to a SIP phone
> (PRI -> SIP). The calling party does not hear ringing and after about five
> seconds gets an *All circuits are busy* recording. However, the called SIP
> phone does ring, and if the called party answers the phone within a few
> seconds, the call stays in service.
>
> CLI messages:
>
> ...
> -- Accepting call from '9288532045' to '6023432727' on channel 0/23,
> span 1
> -- Executing Dial("Zap/23-1", "SIP/102|20|rt") in new stack
> -- Called 102
> !! Don't know how to add an IE High-layer Compatibility (125)
> !! Unable to add IE 'High-layer Compatibility'
> -- SIP/102-935d is ringing
> -- Channel 0/23, span 1 got hangup request
> == Spawn extension (incoming, 6023432727, 1) exited non-zero on 'Zap/23-1'
> -- Hungup 'Zap/23-1'
> ...
>
> NOTE: There is no problem calling from SIP phone out (SIP -> PRI).
>
>
> Any body ever have this problem?
>
> Thanks,
> Murrah
>
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