This is what i found on Cisco's site: "Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message.
Conditions: This symptom is observed on a Cisco platform when the terminating
gateway is configured with the G279B (annex B) codec and when the Session
Description Protocol (SDP) for the incoming Invite message does not have any
FMTP attribute line, which means that the default value, that is, the G279B
(annex B) codec, is used.
Workaround: There is no workaround."
Regards,
Jan
-----Ursprungligt meddelande-----
Från: [EMAIL PROTECTED] genom Gary Richardson
Skickat: on 2006-02-01 21:45
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [Asterisk-Users] Re: CallerID Problem
No, I'm not including the <> -- I was trying to show that it was
something that I removed from my example..
Thanks.
On 2/1/06, Bromont Quebec <[EMAIL PROTECTED]> wrote:
> Are you actually putting the < > in there?
>
> try:
>
> exten => _9.,1,Set(CALLERID(number)=MAINNUMBER)
>
> Hey,
>
> I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
> connects to it using SIP. The asterisk version is 1.2.0.
>
> In my sip.conf, I set callerid="First Last" <exten>
>
> When I make a an outbound call with the following macro:
>
> exten => _9.,1,Dial(SIP/${EXTEN}@<ROUTER>,,w)
> exten => _9.,2,Congestion()
>
> The caller id is set to the extension that's defined in sip.conf.
>
> If I try something like:
>
> exten => _9.,1,Set(CALLERID(number)=<MAINNUMBER>)
> exten => _9.,2,Dial(SIP/${EXTEN}@<ROUTER>,,w)
> exten => _9.,3,Congestion()
>
> I get the following error:
>
> -- Got SIP response 488 "Not Acceptable Media" back from <ROUTER>
>
> It all works fine if I don't set the caller id.. Any ideas on why this
> may be happening?
>
> Thanks.
>
>
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